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Page 1: DSP Modulated Class D Audio Amplifier

DSP Modulated Class D Audio Amplifier

A Major Qualifying Project

Submitted to the Faculty of

WORCESTER POLYTECHNIC INSTITUTE in Partial Fulfillment of the requirements for the

Degree of Bachelor of Science in

Electrical and Computer Engineering

Submitted on April 29, 2009

Sponsoring Agency: New England Center for Analog & Mixed Signal Design

Submitted by: Wadii Bellamine Jameson Collins Craig Ropi Advised by: Professor Andrew Klein

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Abstract

The goal of this project was to create an 80W, 95% efficient Class D audio amplifier with

less than 0.5% harmonic distortion and greater than 100 dB zero-input signal to noise ratio that

accepts digital inputs. The amplifier that was built comprised of a three-state digital modulator, an

H-bridge amplifier, and a passive filter and was capable of accepting both digital and analog audio

inputs by means of the SPDIF protocol and an ADC. To allow the modulator design to be quickly

altered, it was implemented on a DSP. Because the modulator could be easily changed, several

different modulation schemes were simulated, designed and tested in order to achieve optimal audio

quality and efficiency results.

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Table of Contents

ABSTRACT ......................................................................................................... II

TABLE OF CONTENTS ................................................................................. III

LIST OF FIGURES ............................................................................................ VI

LIST OF TABLES .............................................................................................. IX

EXECUTIVE SUMMARY .................................................................................. X

1. INTRODUCTION ............................................................................................1

1.1 Project Goals ................................................................................................................ 2

1.2 Report Organization .................................................................................................... 2

2. BACKGROUND ............................................................................................... 4

2.1 Types of Amplifiers ..................................................................................................... 4

2.1.1 Type A, B, and AB Amplifiers ................................................................................. 4

2.1.2 Class D amplifiers: a conceptual description ................................................................ 7

2.2 Modulation .................................................................................................................... 9

2.2.1 Pulse Width Modulation ............................................................................................ 9

2.2.2 Delta Sigma Modulation .......................................................................................... 10

2.2.3 Signal to Noise Ratio ............................................................................................... 15

2.3 Feedback System Stability ......................................................................................... 15

2.3.1 Phase Margin .......................................................................................................... 16

2.3.2 Increasing System Stability ....................................................................................... 16

2.4 Digital Processing ....................................................................................................... 18

2.4.1 Digital Inputs .......................................................................................................... 18

2.4.2 SNR and Quantization Bits.................................................................................... 19

2.4.3 DSP's ..................................................................................................................... 23

2.4.4 Development Environment........................................................................................ 23

2.4.5 Core Processor Speed ................................................................................................ 25

2.4.6 Floating Point vs. Fixed Point ................................................................................. 27

2.4.7 Digital Outputs ....................................................................................................... 28

2.5 Power ........................................................................................................................... 29

2.5.1 Noise, Distortion, and Filtering ............................................................................... 31

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3 SIMULATION, DESIGN AND PARAMETER SYNTHESIS .................... 34

3.1 Modulator .................................................................................................................... 34

3.1.1 Cascaded Integrator Form ........................................................................................ 35

3.1.2 Loop Filter Form .................................................................................................... 36

3.2 Feedback ..................................................................................................................... 37

3.2.1 Stability Remedies .................................................................................................... 38

3.2.2 Complications .......................................................................................................... 41

3.3 DSP Oriented Simulation ......................................................................................... 43

3.3.1 Software Flow Diagram ........................................................................................... 43

3.3.2 IIR Filter Implementation ........................................................................................ 47

3.3.3 Coefficient Optimization ........................................................................................... 50

3.3.4 SNR Simulation Results ......................................................................................... 52

3.4 DSP Modulator Implementation ............................................................................. 55

3.4.1 System Flow Diagram .............................................................................................. 55

3.4.2 Timing .................................................................................................................... 58

3.4.3 Interrupts and their Pitfalls ...................................................................................... 60

3.5 Hardware and PCB Design ....................................................................................... 61

4 TESTING ........................................................................................................ 64

4.1 Power and Efficiency ................................................................................................. 65

4.2 Input Sources .............................................................................................................. 66

4.3 Audio Quality ............................................................................................................. 66

4.3.1 SNR ....................................................................................................................... 67

4.3.2 THD ...................................................................................................................... 70

4.3.3 Automating Sound Quality Testing .......................................................................... 70

5 RESULTS......................................................................................................... 76

5.1 SNR .............................................................................................................................. 76

5.2 THD ............................................................................................................................ 80

5.3 Efficiency and Power ................................................................................................. 82

6 CONCLUSION ............................................................................................... 87

7 FUTURE WORK AND RECOMMENDATIONS ....................................... 90

REFERENCES .................................................................................................. 92

APPENDIX A CIFB MODULATOR COEFFICIENT GENERATION MATLAB CODE ................................................................................................ 93

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APPENDIX B LOOP FILTER COEFFICIENT GENERATION MATLAB CODE ................................................................................................................. 93

APPENDIX C LOOP FILTER SIMULINK MODELS .................................. 99

APPENDIX D DSP SIMULATION MATLAB CODE .................................. 101

APPENDIX E COEFFICIENT OPTIMIZATION MATLAB CODE ......... 103

APPENDIX F AUTOMATED TESTING MATLAB CODE ........................ 106

APPENDIX G DSP CODE ............................................................................... 111

APPENDIX H RAW ZERO INPUT SNR DATA .......................................... 125

APPENDIX I RAW SNR DATA ...................................................................... 126

APPENDIX J RAW THD DATA ..................................................................... 127

APPENDIX K RAW POWER AND EFFICIENCY DATA ........................... 128

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List of Figures

Figure 1 - Class A Amplifier ............................................................................................................................ 5

Figure 2 - Class B Push-Pull Configuration ................................................................................................... 5

Figure 3 - Class D Stages and Their Frequency Domain Representation ................................................. 7

Figure 4 - (a) Continuous Time ∆� and (b) Linear Z-Domain Model. .................................................. 10

Figure 5 - A Second Order CIFB Modulator Schematic ........................................................................... 11

Figure 6 - The Noise Transfer Function of a Delta Sigma Modulator.................................................... 12

Figure 7 - Loop Filter diagram ...................................................................................................................... 12

Figure 8 - OSR vs. Maximum SQNR for a 2-bit Delta Sigma Modulator .............................................. 14

Figure 9 - SNR and Noise Floor ................................................................................................................... 21

Figure 10 - Common DSP Word Formats and Corresponding SNR ..................................................... 22

Figure 11 - DSP Sampling and Processing Timing .................................................................................... 26

Figure 12 - Digital Output Timing Diagram ............................................................................................... 29

Figure 13 - Half Bridge Power Stage ............................................................................................................ 30

Figure 14 - Full Bridge Power Stage ............................................................................................................. 31

Figure 15 - Example Low Pass Active Filter ............................................................................................... 32

Figure 16 - 2008 Class D Audio MQP Filter Design ................................................................................. 33

Figure 17 - A Second Order CIFB Modulator Schematic ......................................................................... 35

Figure 18 - Simulink Model of 2nd Order CIFB Delta Sigma Modulator.............................................. 36

Figure 19 - Loop Filter diagram .................................................................................................................... 36

Figure 20 - Second order Loop Filter Simulink Model .............................................................................. 37

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Figure 21 - Second Order Modulator with Full System Feedback Simulink Model ............................. 40

Figure 22 - DSPsim() Function ..................................................................................................................... 44

Figure 23 - modulate() Function Block Diagram ....................................................................................... 45

Figure 24 - quantize() Function Block Diagram ......................................................................................... 46

Figure 25 - Tri-state Quantizer Levels ......................................................................................................... 46

Figure 26 - Direct Form II IIR Filter Structure (Source: MATLAB Help) ............................................ 47

Figure 27 - Software Flow Diagram for an N-Order IIR Filter Function .............................................. 49

Figure 28 - dspSim() Output Sample ............................................................................................................ 50

Figure 29 - Average SNR Improvement from Optimized Coefficients for Orders 1-5 ....................... 52

Figure 30: SNR vs. Input Frequency, Simulation ....................................................................................... 53

Figure 31: THD vs. Input Frequency, Simulation ...................................................................................... 54

Figure 32 - Simplified System Flow Diagram.............................................................................................. 56

Figure 33 - Detailed System Flow Diagram................................................................................................. 57

Figure 34 - System Timing Diagram ............................................................................................................. 58

Figure 35 - Timing Relationships .................................................................................................................. 59

Figure 36 - Feedback Capturing Network ................................................................................................... 62

Figure 37 - Class D Audio 2009 PCB ........................................................................................................... 63

Figure 38 - Test Bench (Left to Right: Resistor Test board, Amplifier PCB, DSP) ............................. 64

Figure 39: Flow Chart for Automating SNR and THD Calculations ...................................................... 73

Figure 40 - Zero Input SNR vs. Input Frequency for 1 MHz Modulation Frequency ........................ 77

Figure 41 - Zero Input SNR vs. Input Frequency for 1.5 MHz Modulation Frequency ..................... 77

Figure 42 - Zero Input SNR vs. Input Frequency for 2 MHz Modulation Frequency ........................ 78

Figure 43 - SNR vs. Input Frequency for 1 MHz Modulation Frequency ............................................. 79

Figure 44 - SNR vs. Input Frequency for 1.5 MHz Modulation Frequency .......................................... 79

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Figure 45 - SNR vs. Input Frequency for 2 MHz Modulation Frequency ............................................. 80

Figure 46 - THD vs. Input Frequency for 1 MHz Modulation Frequency ............................................ 81

Figure 47 - THD vs. Input Frequency for 1.5 MHz Modulation Frequency ......................................... 81

Figure 48 - THD vs. Input Frequency for 2 MHz Modulation Frequency ............................................ 82

Figure 49 - Output Power vs. Frequency ..................................................................................................... 83

Figure 50 - Output Power vs. Frequency ..................................................................................................... 84

Figure 51 - Average Efficiency vs. Modulator Order ................................................................................ 84

Figure 52 - Efficiency for All Modulator Orders at 20 kHz ..................................................................... 85

Figure 53 - Average efficiency vs. Modulator Order.................................................................................. 86

Figure 54 - Efficiency of a 4th Order Modulator ....................................................................................... 86

Figure 55 - Efficiency vs. Frequency ............................................................................................................ 87

Figure 56 - First Order Loop Filter Simulink Model ................................................................................. 99

Figure 57 - Second Order Loop Filter Simulink Model ............................................................................ 99

Figure 58 - Third Order Loop Filter Simulink Model ............................................................................. 100

Figure 59 - Forth Order Loop Filter Simulink Model ............................................................................. 100

Figure 60 - Fifth Order Loop Filter Simulink Model ............................................................................... 100

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List of Tables

Table 1 - Project Specifications ....................................................................................................................... 2

Table 2 - Comparison of Amplifiers ............................................................................................................... 7

Table 3 - Dynamic Range of Various Audio Devices ................................................................................ 20

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Executive Summary

Class A, B, and AB amplifiers operate in slightly different ways, but none of them can

achieve power efficiencies above that of the Class D amplifier. The reason for this is that Class D

amplifiers modulate their input in order to make it incredible efficient to amplify the signal. Having

efficiencies this high make Class D amplifiers a perfect fit for any application that demands a

compact size and low power consumption. However, with Class D amplifiers, increasing sound

quality means increasing the complexity of the modulation system. In order to combat this need of

increased complexity, the modulator block for this project was built on a digital signals processor

(DSP). Using a digital signals processor instead of analog components reduced the increased

complexity problem from adding op-amps and flip-flops to simply adding coefficients to an already

existing software based IIR filter. Designing the modulator on a digital signals processor also

allowed for the amplifier to have digital and analog inputs with almost no additional hardware.

Being able to accept a digital input was one of the goals of this project along with achieving 80 watts

of output power, 95% power efficiency, over 100 dB of zero-input SNR and less than 0.5% total

harmonic distortion. This report discusses the design of a Class D amplifier with that means the

previously mentioned goals and all research that was done to build the final working prototype.

The modulation technique chosen for the modulation stage was delta sigma modulation.

This method was chosen over pulse width modulation because of delta sigma’s ability to shape the

modulator’s noise transfer function in such a way that would be advantageous for an audio

application. Instead of choosing a specific modulator order and modulation frequency, we built the

modulator in the DSP in such a way that both could be changed at any given time, until the

processing limits of the DSP were reached. Implementing the modulator with software allowed us

to first build several simulations of the modulator in MATLAB. Eventually a replica of the DSP’s

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code was built in MATLAB to allow our team to troubleshoot and change the code with ease.

These simulations were essential to our project’s success, because they gave our team confidence

that our system would meet our signal quality goals. The modulator was chosen to have a three state

output so that it could correctly interface with the already designed three state power stage.

The power and filter stages chosen for our project were designed and built by the 2008

NECAMSID Class D audio amplifier MQP group. The power stage consisted of a three state

H-bridge amplifier. The filter used was a two pole Butterworth filter. Both the power stage and the

filter were built on one printed circuit board and generously leant to us for the duration of our

project. The reason for the reuse of this portion of the project was the impressive efficiency that

was seen by last year’s design.

One of the major advantages of building the modulator of this project on a DSP was the

ability to switch the order of the modulator as well as its modulation frequency in order to compare

the effectiveness of various combinations. We chose to test modulator orders one through five,

with modulation frequencies of 1 MHz, 1.5 MHz, and 2 MHz for a range of input frequencies in the

audio band. The large amount of testing that was needed forced our team to automate the sound

quality testing process. We did just that by having the DSP internally and automatically create all

test signals and to have MATLAB do all the necessary calculations on the recorded outputs.

The results of the audio quality tests met all of projects goals. The project achieved over 105

dB of zero input SNR and less than 0.5% total harmonic distortion. The project’s power and

efficiency goals were met as well by producing over 90 Watts of output power and consistently

achieving over 95% power efficiency.

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1. Introduction

Audio amplifiers have long been plagued by trade-offs between size, efficiency, and

performance. Traditionally, high performance audio amplifiers have come with large footprints to

make room for their heavy heat sinks. While efficient, low heat amplifiers have been relegated to

portable devices. The reason for this is that in portable devices, the desire for clarity is pushed aside

by the need to retain a small package.

There are several reasons why higher efficiency amplifiers are, and will continue to be in

demand. One reason is the increasing power consumption of home entertainment systems. This is

due to increasing screen size, the increasing number of multimedia accessories and the increasing

cost of energy. However, for many applications the efficiency of an amplifier alone is not enough to

make it competitive. For an audio application, the amplifier must also be able to deliver audio clarity

comparable to less efficient competitors. As more multimedia accessories are added, power usage

can become a concern, second to the concerns of audio quality and a large number of output

channels. At one point stereo outputs were considered luxury, but now it is common to see audio

sources like DVD players and game consoles outputting 6 or 8 channels of audio. These devices

rely primarily on digital audio standards, where several channels can be transmitted over a single

cable.

Class D amplifiers have higher efficiencies when compared to traditional linear amplifiers

because of their switching nature. In Class D operation, an incoming signal is converted to a digital

signal, amplified, and then filtered. Amplification of a digital signal is an efficient process because

the transistors used are not operated in their wasteful triode region. The efficiency of these

amplifiers implies that their heat dissipation is low; heat sinks can be eliminated from the power

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devices, reducing overall system size and cost. Most Class D amplifiers are designed with space,

heat, and power consumption in mind at the expense of audio quality.

Audio quality does not have to suffer if proper care is given in the creation of the amplifier.

High audio quality can be achieved by creating an accurate digital representation of the input signal,

and also by combating the major sources of audio distortion through the use of closed loop full

system feedback.

1.1 Project Goals

This main goal of this project was to create a prototype of a Class D amplifier and present its

function to the NECAMSID lab. The target specifications of the amplifier are shown below in table

1.

Specification Goal

Input Digital and Analog

Output Power 80 Watts

Efficiency > 95%

Zero Input Signal to Noise Ratio >100dB

Total Harmonic Distortion <0.5%

Table 1 - Project Specifications

1.2 Report Organization

This MQP progressed in a different fashion than many other EE projects. Because the

project was based on signals, and digital processing, much of the work was performed in simulation.

The transition from simulation to code is relatively short when compared to the length of the

project, so revisions were created daily. This heavily influenced the organization of the report,

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which does not track the projects progress through revisions, but instead details the process by

which simulation had a daily impact on the design.

This report is organized into several large sections. The first section is background

information. This section discusses the concepts behind Class D amplifiers and modulators, basics

of DSP criteria, and an overview of audio quality standards. The second section discusses the

process by which simulations were created and used to synthesize parameters and fine tune

modulator code. The final two sections discuss the methods by which test data was accumulated

and the results of the testing process.

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2. Background

This chapter supplies an explanation of topics that are necessary in order to understand the

amplifier's design. The topics discussed are types of amplifiers, digital audio inputs, modulation, and

a number of topics from last year's project and report.

2.1 Types of Amplifiers

Before we delve into the intricacies of Class D amplifiers, it is important to explore other

types of amplifiers in order to gain a general understanding of what makes an amplifier more or less

efficient. A comparison between these amplifiers and Class D amplifiers will then allow us to see

why the latter is the best choice for maximizing efficiency. The amplifiers we will explore are the

most commonly found in industry: Class A, B, and AB, although other less common types exist

such as C, E, F, and GH.

2.1.1 Type A, B, and AB Amplifiers

Class A amplifiers are the simplest, but least efficient type. They consist of an output

transistor to amplify the input signal and are therefore conducting 100% of the time, even when

there is no input signal. A representation of a Class A amplifier is given in Figure 1.

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Figure 1 - Class A Amplifier Figure from Wikipedia7

This setup is inherently as linear as possible, allowing for minimal distortion, and therefore

the best audio quality. However, because the transistor is conducting all the time, power is drawn all

the time, and a great amount of heat is dissipated, requiring large and expensive heat sinks for high

power applications. The maximum theoretical efficiency for this type of amplifier is 50% for

inductive output coupling, and 25% for capacitive coupling.

Class B amplifiers are similar to Class A, but each half of the Class B device conducts for

only half the sinusoidal cycle and is off during the second half. Class B devices are usually setup in a

push-pull configuration such that one device conducts during the positive half cycle and the other

for the negative half cycle. The output of both devices is then combined to form a single output.

This behavior is shown in Figure 2 below.

Figure 2 - Class B Push-Pull Configuration

Figure from Wikipedia7

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Because of this setup, Class B amplifiers are more efficient than Class A because if there is

no input signal, there is no current flow at the output. However, the transistors are still dissipating a

significant amount heat when they are conducting, and some linearity is compromised due to the

distortion at the crossover point while switching from one device to the other. The maximum

theoretical efficiency for Class B is 78.5%.

Class AB amplifiers are the same as Class B, except for the fact that each transistor is biased

to allow a portion of the signal to pass after its half cycle is over. This solves the issue of crossover

distortion, but compromises some efficiency by doing so. Class AB amplifiers are therefore more

efficient than Class A, but less efficient that Class B, with a maximum theoretical efficiency of less

than 75.8% (typically 50%).

The amplifiers mentioned thus far are all linear amplifiers. Class D amplifiers are of a

different nature: they take advantage of switching to maximize efficiency. Class D amplifiers

modulate the input signal to convert it into a pulse code format which consists of two values: high

and low. This allows Class D amplifiers to take advantage of MOSFETS which act as switches,

allowing or disallowing the passage of power to the output stage. Because MOSFETS consume very

little power when operating in their triode regions (which are the only regions they need to operate

at due to the pulse code signal at their gate), very high efficiencies are possible. Class D amplifiers

are able to attain over 95% efficiency. This, in turn, means that virtually no heat sinks are required,

reducing the size and cost of Class D amplifiers. Nevertheless, Class D amplifiers suffer from

higher distortion than the linear amplifiers mentioned before, because of the noise introduced by the

switching, the quantization of the linear input signal, and possible phase delays introduced at every

stage. For this reason, Class D amplifiers tend to be more complicated because additional circuitry

is required to compensate for distortion.

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The following table summarizes and compares the power amplifier types mentioned in this

section:

Table 2 - Comparison of Amplifiers

2.1.2 Class D amplifiers: a conceptual description

Class D amplifiers consist of three stages: a modulation stage, a power stage, and a

demodulation stage. With this in mind, consider the following figure:

Figure 3 - Class D Stages and Their Frequency Domain Representation

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Let us assume for now that the amplifier takes a continuous-time analog input signal. Let us

then suppose that this input is a simple sine wave, which will have a single frequency, as shown in

the leftmost frequency plot. The modulation stage's role is to convert this signal into a digital pulse

code, which in the model above is a two-state pulse code. There are several modulation techniques

that can be used to accomplish this, but we only need to quantize the signal in order to represent it

as a square wave with pulses of varying densities. The modulation stage therefore adds two types of

high frequency noise: one is the noise produced by the quantization of the input signal, and the

other is inherent from the frequency spectrum of any square wave. A square wave is made up of a

fundamental frequency and all odd harmonics, theoretically to infinity. This explains why the

frequency spectrum shown in the middle of Figure 3 consists of the fundamental frequency of the

input signal, along with a range of additional high frequencies.

The power stage then takes this pulse code output from the modulator and amplifies it to

the desired power. Ideally, this will only increase the magnitudes of all frequencies if we look at the

frequency representation of the output of the power stage. However, the quality and switching

speed of the MOSFETS used in the power stage is not perfect, and may add additional high

frequency noise.

The final step is to remove all the added high frequency noise, which is accomplished by a

low-pass filter. The low-pass filter attenuates all undesired frequencies, but leaves all the frequencies

in the pass-band untouched. Ideally, if all added frequencies were outside the audible range, only the

input’s frequencies would remain. In the example above, the single frequency of the original signal

remains, so an amplified version of the input is received at the output.

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2.2 Modulation

The modulation stage of a Class D amplifier is a critical block with respect to audio quality,

because the modulation stage can induce large amounts of noise and harmonic distortion if not

designed carefully. At this stage, the choices regarding the over sampling ratio and the order of the

modulator determine the effective number of bits created by the modulator, as well as the expected

signal to noise ratio. Care must be taken during the design of the modulator to ensure that the

digital signal that is created contains the correct frequency information from the input, and that

there is minimal harmonic distortion and attenuation. The modulation stage must also be designed

with the proposed power stage in mind. Sampling frequencies of the modulator need to be high

enough to shape noise out of the signal band but low enough to remain in the efficient regions of

the power components.

2.2.1 Pulse Width Modulation

In pulse width modulation (PWM), the input signal is represented by a digital data stream

where the duty cycle of the output signal is proportional to the input signal amplitude. There are

various ways to implement this modulation technique but the most common method uses a high

speed comparator and a ramp signal that acts as a carrier for the output. The input signal is

compared to the ramp; the output of the comparator is the digital signal. The frequency of the

carrier signal needs to be at least twice that of the input signal, but a typical carrier frequency is

around 500 kHz.

While PWM is easy to implement, is used in a variety of systems, highly commercially

available, and offers excellent SNR, it has serious draw backs that limit its use in a high fidelity audio

amplifier. The noise in the modulated signal has high power in narrow frequency ranges, such as the

carrier frequency. Removing these bands of high power noise requires a complex filtering stage.

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2.2.2 Delta Sigma Modulation

Delta Sigma modulation uses a technique known as pulse density modulation to encode high

resolution data into a low resolution bit stream (the modulated signal). Pulse density modulation

represents the amplitude of an analog signal with the relative density of the output pulses. In a

continuous time representation of a delta sigma modulator, an input signal is first integrated. At the

sampling frequency, Fs, the integration is compared to a predetermined threshold, which either flips

or resets the output. Because the output can only be changed at interval 1/Fs the width of the

digital pulses are always the same. Even in the most basic Delta Sigma modulators, closed loop

feedback is used to monitor the accuracy of the output. The continuous and discrete time models of

a first order Delta Sigma modulator are depicted below in Figure 4.

Figure 4 - (a) Continuous Time ∆� and (b) Linear Z-Domain Model.

When higher order delta sigma modulators are built, many scale factors can be added in

order to manipulate the modulator's performance. A Sample modulator structure, called Cascade-

of-integrators, feedback form (CIFB), is shown in Figure 5. In this diagram, z is the discrete time

variable z, n is the order of the modulator, and all a's, b's, c's, and g's are constant gains. The

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"integrators" seen in this model are actually integrators with inherent delays. Basically, each 1/(z-1)

block contains a discrete time integrator preceded by a one unit delay. This form was one of the

ones used in the process of designing our final modulator.

Figure 5 - A Second Order CIFB Modulator Schematic Figure From Schreier, R.5

Delta sigma modulation modeling is made easier by replacing the non-linear quantizer with

an additive error signal E(z). This error signal is the difference between the integrated signal and the

modulated signal. From this model we can obtain the formula V(z) = [z-1 • U(z)] + [(1-z-1) • E(z)].

This equation can be written in terms of a noise transfer function (NTF) and a signal transfer

function (STF). The STF of the modulator defines the frequency region which the modulator will

pass from input to output. A modulator used for audio will have an STF of 1 or it will be a low

pass filter surrounding the audio band. The NTF defines the region that quantization noise will

exist. For an audio modulator the NTF will be a high pass filter. An effective modulator will

separate the NTF and STF as far as possible to ensure quantization noise does not over run the

signal band. A plot of the NTF of a first order modulator is depicted in Figure 6.

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Figure 6 - The Noise Transfer Function of a Delta Sigma Modulator

If the NTF and STF of a modulator are known, the modulator can be turned into a single

loop filter with two filters G and H, as shown below in Figure 7. In this diagram, G = ������ and

H = �������� , which can be confirmed by solving the output in terms of the input and the error. Both

G and H filters are the same order as the order of the delta sigma modulator that they produce.

Figure 7 - Loop Filter diagram

Two parameters govern the effectiveness of a Delta Sigma modulator; those are the OSR

(over sampling ratio) and the order of delta sigma modulator. In continuous time models, the order

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of the modulator is directly related to the number of integrators in the signal path. In a discrete time

modulator, the order of the system is the order of the NTF. A higher order NTF increased the

slope of the cutoff of the filter, decreasing the noise in the signal band. The over-sampling ratio

(OSR) is OSR =�

��� , where Fb is the bandwidth of the signal and Fs is the sampling frequency of

the modulator. Increasing the OSR shifts the NTF out of the signal band. From figure 7 we see in

a discrete time system that the noise filter is a function of the normalized frequency. When the OSR

is high the signal band moves lower in the normalized frequency. The expected SNR of these

systems can be estimated from the order of the system and the OSR (assuming the OSR >> 1,

typically 22 to 210). In order to achieve a specific SNR, there is a trade-off between the order of the

modulator and the OSR needed. Figure 8 below shows the trade-off between OSR and maximum

SNR for a 2-bit Nth order delta sigma modulator. In this figure, SNR is shown as SQNR or Signal

to Quantization Noise Ratio. For our purposes, these terms are synonymous.

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Figure 8 - OSR vs. Maximum SQNR for a 2-bit Delta Sigma Modulator Figure from Schreier5

As stated earlier, delta sigma modulation can offer superior SNR because of its internal

feedback and its ability to shape the noise out of the signal band. This fact makes the delta sigma

modulator a good choice for an amplifier where high audio quality is required. Since the quality of

the delta sigma modulator is dependent on both OSR and the order of the system being high, and a

feedback loop required for the system, the stability of the modulator must be considered. Increasing

the order of the modulator increases the phase delay of the system, which could potentially

destabilize the modulator for some or all input amplitudes.

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2.2.3 Signal to Noise Ratio

Signal to Noise Ratio is defined as shown in Equation 1. This means that the SNR of the

system can only be improved by increasing the power of the desired signal or by reducing the power

of the noise.

�� � �������������� ���������� �����

Equation 1 - Definition of SNR

In the previous sections we introduced the NTF of a modulator. We explained that

increasing the order of the modulator increases the order of the NTF. Increasing the modulation

frequency moves the NTF further out of the signal band. Moving the NTF out of the signal band

increases the SNR of the system. An alternate way to categorize the modulator's SNR is by using its

effective number of bits (ENOB). As mentioned in previous sections, the number of bits used in an

analog to digital converter is directly related to its SNR. With a delta sigma modulator we are limited

to a low number of output states to remain compatible with a Class D power stage. In a delta sigma

modulator we can compensate for less output bits with a higher modulation frequency. By keeping

the modulation frequency high we can maintain a high ENOB and therefore maintain a high SNR

even with a small number of output bits.

2.3 Feedback System Stability

Having full system feedback in a Class D amplifier system has the potential to greatly

decrease both noise and distortion. Like in any other feedback system, the output can be monitored

and checked against the input in order to eliminate noise or distortion that is added by the system

itself. One of the main challenges involved in building a system with feedback is making sure it is

stable for all expected inputs. According to Dorf(1989), Feedback systems are considered stable if

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for any bounded input, they produce a bounded output. One way to determine system stability is by

referencing the phase margin of a system. There are also many circuits which can improve the

stability of a system, three of which are named phase-lead, phase-lag, and lead-lag networks.

2.3.1 Phase Margin

Phase margin is a widely used measure of system stability when working in the frequency

domain. Phase margin is defined as the distance of the system transfer function’s phase shift from

180º when at the system’s zero dB point or the frequency at which the system’s amplification factor

is 1. The phase margin's positive distance from zero can determine the stability of a system. This

makes sense, because a system inherently oscillates when its gain is one and its phase shift is 180º.

When a system's phase margin is below zero, the system is considered unstable, because its output

could approach infinity. Phase margin can be determined using Nyquist plots such as the ones

explained in the previous section, but it can also be measured using Bode plots. On a system's Bode

plots, the phase margin represents the frequency response's distance from -180˚ at the point where

the system has zero gain. This point is known as the zero dB point.

2.3.2 Increasing System Stability

There are several approaches to making a system more stable. Many of them are rooted in

changing the design characteristics of the functional blocks of the system. If this sort of change is

not possible, or is not enough to make the system stable, another approach can be taken. This

approach is to add another block to the feedback loop, GC(s). This function is called a

compensator, because it compensates for the system's previous inadequacies.

One type of compensator is the phase-lead network. This block is made simply from a resistor

which is in parallel with a resistor and capacitor and has the transfer function given below in

Equation 2. In this equation, R1 is the resistor in parallel with the parallel combination of R2 and C.

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!"#$ � %

% & ' %� � 1)# *%� & � 1)# +

,

Equation 2 - Transfer Function of Phase-Lead Network

This block also has a positive phase angle of almost 90 degrees. Since blocks in series simply

add their phase responses, this block's positive phase response can directly increase the phase

margin of the entire system. Another side effect of the phase lead function is that it is an amplifier

for frequencies greater than the location of its own zero. This could distort the audio signal in our

project if the necessary frequency of this block's zero is lower than 20 kHz.

Another type of compensator is the phase-lag network. This block is made from the same

components as the phase-lead, with them arranged in a different way. For the phase-lag network,

there is a resistor in parallel with a series combination of another resistor and a capacitor. The

transfer function for this block is given below in Equation 3. For this equation, the R1 is the resistor

in parallel with the series combination of C and R2.

!"#$ � % & � 1)# %� & % & � 1)#

Equation 3 - Transfer Function of Phase-Lag Network

This block actually has a negative phase response at a very low frequency. This should not

affect the phase margin of the system, because the frequencies at which the phase-lag network's

phase response is negative should be much lower than the crossover frequency of the system.

Instead of directly affecting the phase margin of the system, the phase-lag network adds attenuation,

moving the crossover frequency to a much smaller frequency. At this new frequency the phase

margin will be greater and therefore the system will be more stable. Because this block attenuates

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some parts of the audio band of frequencies more than others, no analog audio signal can be passed

through it without being distorted.

2.4 Digital Processing

The core of our project was the digital modulator. A digital approach implies a range of

design considerations that are different or not needed in an analog approach. Questions regarding

the different types of digital input standards and connectors, and quantization bits and its effects on

SNR, need to be answered prior to beginning the design. More specific DSP-related issues, such as

the required features of the development environment, floating point versus fixed point calculations,

core processor speed, and digital output capability, are all addressed along with the previous

questions in this section.

2.4.1 Digital Inputs

The consumer digital audio market is dominated by a standard know as S/PDIF

(Sony/Philips Digital Interconnect Format). The standardized name for SPDIF is IE60958-3, also

referred to as AES/EBU where it is known as IE60958 Type II. Type I uses a balanced 110-ohm

twisted pair wire with an XLR head, this setup is used in professional audio installations where the

XLR connector is commonly used for many mic level sources.

Type II comes in two formats, unbalanced and optical. The unbalanced signal is transmitted

on a 75-ohm coaxial cable with an RCA head. This wire is generally utilized in consumer audio

applications where the RCA connector is already commonly used for both mic and line level sources

as well as many video sources. The second format of type II uses an optical cable, consisting of

plastic or glass with an F05 (or trade name TOSLINK) connector. In consumer applications the

F05 optical connector has been limited to transmitting digital audio.

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There is no difference between the signals transmitted with either cable connector

combination. Selection of either connector relies primarily on availability and consumer preference.

On a more technical level, for connections longer than 6 meters or where tight bends are required,

coaxial cables should be used because of the attenuation of the fiber wires is rather large and limits

the effective range. However, if distance and routing is not an issue, the optical wire is superior

because it is impervious to distortion and noise caused by RF interference and ground loop issues. A

final deciding factor may be cost. The ubiquitous 75-ohm coaxial cable with RCA connector is

given away with nearly every piece of audio equipment, while the fiber cable is rarer and therefore

more expensive.

SPDIF can be used to transmit a number of different formats. The two most common

formats being the 48 kHz signal used in DAT (digital audio tape) and 44.1 kHz used in CD audio. In

order to support a full range of formats the SPDIF protocol has no defined data rate. The SPDIF

protocol uses BMC (bi-phase mark code) which encodes the original word clock into the data

stream by transmitting two bits to represent a signal data bit, allowing it to be recreated by the

receiver. A 48 kHz sample rate results in a bit rate of 3.072 MHz and a clock rate of 6.144 MHz

SPDIF transmits 32 bit data streams of PCM encoded audio. The audio signal is divided into

blocks, frames and sub frames. A sub-frame is a single audio sample, including some header data,

channel data, parity data, etc. A frame contains a single sub-frame from every channel in the

system. A block is made up of frames, the block structure is used so that additional data can be

transmitted at the end of each sub-frame.

2.4.2 SNR and Quantization Bits

The core of our Class D amplifier is the modulation which is performed in the DSP. The

nature of this modulation is therefore digital. Whether the input to the modulator is analog or

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digital, it will have to be converted, using an ADC, into a digital sample. This means that the input

signal is quantized, and the sample represented using an n-bit value. Our project goals require over

100dB SNR, and it is important to understand the relationship between quantization and how it can

limit our maximum attainable SNR. In this section we try to understand this relationship, and

determine the minimum bits needed to represent the sample such that our SNR is not limited below

100dB.

In analog terms, SNR refers to the ratio of the largest known signal to the noise present

when no signal exists1. In digital terms, SNR and dynamic range are used interchangeably to refer to

the ratio between the largest number that can be represented and the quantization error or

quantization step. Higher SNR means a higher audio signal quality, as the following table indicates:

Audio Device/Application Dynamic Range

AM Radio 48 dB Analog Broadcast TV 60 dB FM Radio 70 dB Analog Cassette Player 73 dB Video Camcorder 75 dB ADI SoundPort Codecs 80 dB 16-bit Audio Converters 90 to 95 dB Digital Broadcast TV 85 dB Mini-Disk Player 90 dB CD Player 92 to 96 dB 18-bit Audio Converters 104 dB Digital Audio Tape (DAT) 110 dB 20-bit Audio Converters 110 dB 24-bit Audio Converters 110 to 120 dB Analog Microphone 120 dB Table 3 - Dynamic Range of Various Audio Devices

Figure from Tomarakos, J.6

1 Another common definition for SNR is the ratio of the power of the noise of a signal to the power of the signal without the noise.

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In an ADC, the difference between two consecutive binary values is known as the quantization step

or quantization level. The size of the step defines the noise floor. The noise floor is the point

where the audio signal cannot be distinguished from low-level white noise. These terms can be

visualized as follows:

Figure 9 - SNR and Noise Floor Figure from Tomarakos, J.6

Simply put, the higher the number of bits, the smaller the quantization step, and therefore

the higher the SNR. The formula for the SNR of an ADC, where n is the number of bits, is given

as:

SNR ADC(RMS) [dB] = 6.02n +1.76 dB

Note that 1.76dB is based on sine wave statistics and can vary for other types of waveforms.

For large enough n values, 1.76dB can be ignored and the SNR is estimated as 6n dB, or an

additional 6 dB SNR for every additional bit (Tomarakos). The bottom line is that any ADC used in

this amplifier must have enough bits to produce the desired signal quality (or SNR or dynamic

range). The following diagram illustrates the available DSP types and their corresponding SNR's:

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Figure 10 - Common DSP Word Formats and Corresponding SNR Figure from Tomarakos, J.6

In conclusion, in order for our SNR to not be limited by any ADC used, we must choose at

least a 24-bit ADC and 24bit words.

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2.4.3 DSP's

Choosing the right DSP is an important step that can either hinder or facilitate our ability to

perform proper analysis, testing, and debugging during the design process. This DSP must have an

adequate development environment with live debugging features, and a rich interface. Other factors

such as floating point vs. fixed point processors, digital inputs, digital outputs, and core processor

speed must meet certain minimum criteria such that our desired project objectives are not limited by

the DSP itself.

2.4.4 Development Environment

The chosen DSP must be compatible with a development environment (or IDE) that offers

three main features: live debugging, statistical profiling, and optimization. The first is necessary to

facilitate and speed up coding. Let us consider our approach to implementing a design on the DSP.

First, we visually design the modulator in MATLAB's Simulink, using blocks for IIR filters, adders,

delays, inputs, and outputs. Once the design is shown to be functional, we then translate these

blocks into MATLAB code and test it once again. This offers the advantage of being able to

generate a signal, pass it to our MATLAB program as an array of data, and flexibly view the results

in formatted plots. After the plots show that we obtain the desired results, we then move on to the

final step, which is to translate the MATLAB code into DSP code. It is in this step where live

debugging becomes important. The major difference between the MATLAB program and the DSP

program is that in the DSP, input is not obtained via a predefined buffer of data. Rather, it is

sampled at a certain frequency, and is interrupt-driven. This means that some changes must be

made to the original MATLAB code to make this possible. It also means that new problems may

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arise while making these changes. Without a robust and non-buggy debugging feature, a rather

trivial problem may not be apparent in the code, and the results become confusing and unexpected,

and can take significant time to sort out. Live debugging allows us to step through the code, and

verify whether or not each step produces the expected result. When the variables differ from what

is expected, we can easily determine, based on the break-point of the program, what may have

caused the error. As a note, the ability to easily watch variables in real time is also a desirable IDE

feature.

As aforementioned, the DSP obtains samples at a certain frequency. Whenever a new

sample is ready, an interrupt is generated, and triggers the predefined interrupt service routine (ISR).

This offers the ability to process a signal in real time, given that the code in the ISR completes its

function before the next sample arrives. Indeed, this poses a strict time limit to the code in the ISR,

which we must take into account when coding our modulator. Without some kind of statistical

profiling available in the IDE or provided with the IDE's libraries, measuring the time taken to

execute the code in our ISR is a difficult task which involves approximating the number of CPU

instructions our code will generate to calculate the total time given the time taken per instruction.

When different types of instructions vary in execution time, and if the code is written in c, the task is

even more complex. The most efficient way to measure execution time is to start a timer prior to

the first instruction in the ISR, and stop it after the last instruction, then convert the timer result and

display it in seconds. The IDE we used did offer such a prebuilt timing function, which was helpful

in indicating whether or not the code took no longer than the allowed time. Statistical profiling,

however, also involves the ability to look at the data being read or generated by the DSP in a visual

way. Buffering a time-varying variable (such as the input), and then displaying the buffered data in a

plot within the IDE can save us the hassle of having to export the data and plot it in a third-party

software. The ability to perform FFT and other analysis functions within the IDE is also a plus

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The last of the three pillars of IDE requirements is optimization. Given that we lack the

time or expertise to write a fully pipelined and optimized assembly code to perform our modulation

function, our best option is to do it in C, which, needless to say, requires an IDE that can compile C

code. However, C code is not always compiled in an efficient way. In fact, previous experience has

shown that a non-optimized C code can take three times as many cycles to run as a sub-optimized

assembly code would. After optimization, however, the C code became twice as fast as the assembly

code, indicating a potential six-fold gain in efficiency post-optimization. The IDE's ability to

provide optimization for C code is therefore important because execution time is of the essence.

Having the tools to measure the execution time per interrupt, and ways to increase processing speed,

the question then becomes what is the maximum execution time per interrupt that is allowed by the

DSP, how do we calculate it, and what limitations does it imply?

2.4.5 Core Processor Speed

It is easy and obvious to say that the faster the CPU is, the easier it is for us to implement

higher order modulators on the DSP. However, it takes some analysis to determine a reasonable

constraint for the minimum core CPU speed required. Consider the following diagram:

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Figure 11 - DSP Sampling and Processing Timing

The bold lines indicate the samples obtained from the DSP's input, which are separated by

tISR , or one over the sampling frequency. Moore's law tells us that a delta sigma modulator must

output at a frequency greater than or equal to 60 times the highest frequency. In our case, the

highest frequency in the audio band is 20 kHz, so 60 * 20 kHz = 1.2 MHz. The samples between

the bold lines represent the samples that will be output from the DSP, and these must be output at a

frequency greater than or equal to the modulation frequency (1.2 MHz). This gives us tMOD = 1/

fMOD amount of time to complete a full modulation cycle. If we let tCOMPU represent the time taken

to complete the computation of a full modulation cycle, tCOMPU must be smaller than tMOD to allow

for a small margin of time, tMGN , which will be used to wait for the next computation to begin. This

defines our major constraint, which is broken down as follows:

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The frequency of the core CPU must therefore be greater than the number of instructions per ISR

(n) times the modulation frequency. For fMOD = 1.2 MHz, and an estimate of 300 instructions per

ISR, our minimum core clock frequency must be greater than 360 Mhz.

2.4.6 Floating Point vs. Fixed Point

Knowing that we are constrained in time, and that we want to minimize the number of

instructions per clock cycle, while not compromising sample integrity, we must face the question of

whether to use fixed point or floating point calculations in our DSP. In an n-bit processor, fixed

point numbers can be represented as signed or unsigned integers or fractionals. A signed integer

ranges from 0 to 2n , unsigned ranges from -2n-1 to 2n-1 and all values within these ranges are equally

spaced. Fractionals are normalized to range between -1 and 1 (for signed) or 0 and 1

(unsigned), which is done by dividing the integer value by 2n-1 (for signed). Floating point numbers,

on the other hand, are represented as scientific numbers (IEEE standard), and for 32bit floats, the

largest and smallest numbers are ±3.4×1038 and ±1.2×10-38 respectively. Floats offer the advantage

of being able to represent much larger and much smaller numbers than fixed point, by

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compromising the equality of distribution of numbers within that range. That is, large gaps separate

two consecutive large numbers, and small gaps separate two consecutive small numbers.

In many DSP's, fixed point is known to be more efficient than floating point because it

involves less operations and usually less bits. However, SHARC processors are designed to be

equally efficient for both fixed point and floating point calculations. Setting that note aside, there is

a major disadvantage in using fixed point numbers: the ease at which overflow can occur. If a

calculated number turns out to be greater than the largest possible fixed point value, then we now

have more bits than can be stored in the register containing the fixed point value. This produces

unpredictable results, and is usually handled by pre-scaling the operands used to calculate the result

such that the result does not overflow. Fixed point calculations, on the other hand, can store very

large numbers while still being able to represent very small numbers. In the scenario where we are

working with an input that ranges from -1.0 to 1.0, calculations can yield our desired range with high

precision, while allowing results to be much greater than our range. Considering the fact that no

efficiency is lost when using floating point in a SHARC processor, one can conclude that floating

point is the best choice if we do use such a processor.

2.4.7 Digital Outputs

Another important consideration is the ability to output a high frequency PDM signal from

our DSP with minimum distortion. We also need at least two digital outputs in order to produce a

tri-state signal. These outputs must be able to switch at frequency greater than or equal to the

minimum modulation frequency, or 1.2 MHz. This means the rise time or fall time of the switching,

plus the response time (the time taken between the instruction to switch and the actual start of the

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switch) must both be smaller than or equal to 1/fMOD. The following diagram illustrates this

concept:

Figure 12 - Digital Output Timing Diagram

tRSP : response time

tLH : Transition time form low to high

tHL : Transition time from high to low

Every switch in state is preceded by a response time tRSP which varies depending on the DSP's

core speed and architecture. In short, the DSP must have 2 digital outputs, where for each of these,

tLH + tRSP and tHL + tRSP are less than or equal to tMOD.

2.5 Power

The power stage of a Class D amplifier accepts a modulated signal, amplifies it and delivers

it to the filter and speaker. Because a Class D power stage is amplifying a digital signal it need only

provide full power or no power. This means the transistors used are either on or off, and do not

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have to operate in their inefficient linear regions. The two most common configurations for a Class

D power stage are H-bridge and half bridge.

In a half bridge configuration there are two transistors. One transistor is attached to the

positive rail and the other is attached to the negative rail. The load (speaker) has one lead connected

between these transistors and the other lead is attached to a bias voltage equal to half the difference

between the power rails. Only one transistor can be on at a time, otherwise the voltage rails would

short together. Enabling one of the transistors supplies the load with a positive voltage, enabling

the other provides a negative voltage. Several issues exist with the half bridge configuration. This

power stage is only capable of two states: positive and negative. At idle, the load will be biased

above ground. Also the maximum output voltage can only be half the power supply voltage.

Figure 13 - Half Bridge Power Stage

In an H bridge configuration there are four transistors. Two transistors are placed in series

between the upper and lower power rails, and another pair of transistors are also placed in the same

configuration. The load is attached between the transistors of these pairs. This configuration is

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capable of three states: positive, negative, and neutral. This is accomplished by changing the polarity

of the speaker leads to create the positive and negative voltage swings, and also by grounding both

sides of the load to create a zero state. Because this configuration only changes the polarity of the

speakers connection, it is capable of outputting the full power supply voltage to the speaker, as

opposed to half the supply as in the half bridge configuration.

Figure 14 - Full Bridge Power Stage

The Class D MQP team from 2008 designed and built a very efficient H bridge power stage.

In their testing it achieved over 95% efficiency. Because the goal of this year’s MQP is the addition

of digital inputs, digitally controlled modulation, and feedback control to the Class D amplifier and

because the 2008 team was so successful with their power stage, it was decided that their power

stage design would be reused.

2.5.1 Noise, Distortion, and Filtering

The goal of a Class D amplifier is to convert a continuous time waveform into a series of

binary pulses so they may be more efficiently amplified. This conversion adds unwanted harmonics

as well as other noises to the output. Some of these are audible, and therefore undesired for noise

quality reasons. Other noises are inaudible, but still undesired for efficiency reasons.

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A Class D modulator runs at speeds significantly higher than the highest desired output

frequency, as dictated by the over sampling ratio. The noise introduced by the modulator is

intentionally "shaped" out of the audio band. Despite the audio quality benefits of this "shaping", if

the constant switching of the modulation stage was allowed to propagate to the speaker the overall

efficiency of the system would be decreased. Although the noise added by the modulation stage is

in audible, it still requires power to push the speaker cone. To increase the efficiency of the system,

and reduce the possibility of cone damage because of high frequency oscillation, the output is

filtered to remove this high frequency noise.

Not all noise added by the modulator will be outside of the audio band. Low fundamental

frequencies will result in odd order harmonics that may lie within the audio band, increasing THD.

In band noise cannot be filtered at the output.

Low pass filters can be implemented either actively or passively. Active filters exhibit

exceptional frequency roll offs, and have the ability to have extremely minimal pass band rippling.

Active filters do not rely on non-ideal components like inductors and capacitors, whose non-linearity

will distort the filter output.

Figure 15 - Example Low Pass Active Filter

Unfortunately the basic design of an active filter makes it impractical for Class D amplifiers.

Active filters are based on amplifiers themselves. Class D amplifiers are efficient because of their

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switching power supplies, the addition of a linear amplifier would defeat the purposes of making a

high efficiency amplifier.

A passive filter is a more practical choice for a Class D amplifier. A passive filter can, at

best, have unity gain. A passive Butterworth filter has a very flat pass-band, although its frequency

cutoff is not as sharp as some other filter configurations. Passive filters are built with non-linear

components, producing output distortion. Last year's team created a passive Butterworth output

filter that performed as required, and this design was reused for this year. The following figure

shows the balanced low pass filter configuration and component values used by last year’s team.

The component values used result in a 30 kHz cut off.

Figure 16 - 2008 Class D Audio MQP Filter Design

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3 Simulation, Design and Parameter Synthesis

The design of our modulator and feedback systems were completed as iterative simulation

processes. This allowed us to constantly edit major portions of design until the day we began

testing. This also allows for future work to easily improve on the progress we have made, including

continuing our research on full system feedback. We were able to take this approach to design

because the modulator and feedback systems were implemented with software on a DSP.

The PCB design which was created with the intention of allowing our system to have

feedback was only designed once. While much of this PCB design was recycled from last year’s

award winning project, only the newly designed portion of our PCB functioned correctly. In this

chapter, the design of the modulator in both MATLAB and a digital signals processor will be

discussed alongside the feedback design created in Simulink and the hardware PCB design.

3.1 Modulator

In order to design a modulator fit for a DSP, several different models were built in

MATLAB before finally porting the design onto a digital signals processor. While researching and

defining the general structure of the modulator, models were built in Simulink, MATLAB’s graphical

model based design simulator. There were two major modulator structures that were built and

simulated; the cascaded integrator model and the loop filter model. The loop filter design was

eventually chosen because of its portability to a DSP.

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3.1.1 Cascaded Integrator Form

There are several different delta sigma modulator structures that can be used to realize any

given noise and system transfer function pair. One such model is called the cascade of integrators,

feedback (CIFB) structure. This modulator topology is examined in detail in section 2.2.2, the delta-

sigma modulation section of the background chapter. The second order form of the modulator

structure is also shown in Figure 17 below for convenience.

Figure 17 - A Second Order CIFB Modulator Schematic Figure from Schreier5

In order to determine the value of all the constants to be used in the simulation, the

MATLAB code in Appendix A was used. Running this code for a second order modulator with an

OSR of 50 and plugging the resulting coefficients into the general CIFB structure resulted in the

Simulink model shown below in Figure 18. Notice that the discrete time integrators in Figure 18 do

not differ from the integrators in the general structure; they are simply represented with z-1 instead

of z. Also notice that the variable g1 is given as zero by the MATLAB code in the appendix, so that

portion of the modulator is omitted. There is a two pole Butterworth filter on the output of the

modulator in order to ensure that the input sine wave can be reproduced.

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Figure 18 - Simulink Model of 2nd Order CIFB Delta Sigma Modulator

3.1.2 Loop Filter Form

In the background section 2.2.2 of this report, it was shown that a loop filter could represent

any modulator topology. This modulator form has a much simpler appearance and is easily

represented by C code in a DSP. The reason this form can be easily implemented in a DSP is the

fact that it consists of only IIR filters. The process of actually implementing an IIR filter in a DSP is

discussed in section 3.3.2 IIR Filter Implementation of this report. The general loop filter form is

shown below in Figure 19 for convenience.

Figure 19 - Loop Filter diagram

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In order to convert an NTF and STF pair into a general loop filter form, the MATLAB code

in Appendix B was created. This code takes an NTF created by the delta sigma toolbox for a certain

order modulator, and creates the two loop filter blocks, G and H, while assuming an STF of 1. The

Simulink model shown in Figure 20 was created after running the script in appendix B and placing

the resulting G and H blocks into the general loop filter model.

Figure 20 - Second order Loop Filter Simulink Model

In this model, the error added to the output is replaced by a two bit quantizer. Once again, a

2 pole Butterworth filter is attached to the output in order to test if the input sine wave can be re-

created from the output of the modulator. Only the second order model is shown here, but models

for orders 1-5 can be seen in Appendix C.

3.2 Feedback

Feedback can theoretically reduce both the noise and distortion of almost any system.

Feedback can also make the same system unstable if it is implemented without great attention to

delays and phase shifts within the system. When conceptualizing our amplifier with full system

feedback, there are two blocks that attribute significant phase shift. The first is the two pole

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Butterworth filter used to low-pass our output just before placing it across the speaker. This filter

adds more than 80º phase shift to the system before the end of the audio band.

This second source of phase shift comes from the delay in the analog to digital converter

going into the DSP that executes the modulation. The output of the amplifier is fed back as an

analog signal, but must be passed into the DSP in order for it to be used in future modulation

iterations. Therefore, it must be passed through an ADC, all of which have a relatively long delay

called a group delay. For the ADC we had available, the time between when the analog signal first

enters the ADC and when a digital representation of that voltage is available at the output is 460µs.

Because of the phase shift induced by a delay is 360*f*d, even a 1 KHz signal would have 165.6º

phase shift from the group delay alone.

Since the audio band contains information up to 20 kHz, phase shift induced by the group

delay would be up to 3312º. This would make the system wildly unstable if feedback were

implemented this way. There are several things that can be done to avoid these stability problems,

while still enjoying the benefits of full system feedback.

3.2.1 Stability Remedies

While designing implementations of feedback for our amplifier, two techniques were used to

combat the crippling phase shift that was added to higher frequencies by the ADC's group delay.

Phase shift is directly related to delay by the following relationship: phase shift = 360*f*d. By

feeding back only the lower frequency band of the output, the phase shift can be reduced to a

minimal and stable value. In order to only feedback lower frequencies, a low pass filter can be

added to the feedback network. Although this compromise would greatly increase stability, it would

also remove any of the benefits of feedback for the higher frequency bands.

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In order to actually implement this idea into our Simulink models, it was decided that two

modulators could be used instead of one. The first modulator would be exactly the same as the

original, but the second modulator's H-block would take its feedback from an attenuated, low pass

filtered, and delayed version of the amplifier's output. A Simulink model of the system that was

described is shown below in Figure 21. Notice that in Figure 21, the first modulator is represented

by G1 and H1, while the second modulator is represented by G2 and H2. In this model, the LPF

used to eliminate higher frequencies from the feedback path is created by the 1 kHz feedback filter

and the inverted version of that filter. Having these two filters where they are adds a 1 kHz LPF to

the signal transfer function of the second modulator.

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Figure 21 - Second Order Modulator with Full System Feedback Simulink Model

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3.2.2 Complications

After building the Simulink model shown in Figure 21, it became clear that optimizing a

system with feedback required a huge amount of research on the topic. After building the model,

several parameters, such as the gain labeled "gain" and the coefficients of the second modulator

were fairly arbitrarily chosen in order to test the prototyped feedback model. The system produced

a completely stable output with the same shape of the input. The model produced similar quality

outputs for inputs both above and below the 1 kHz feedback cutoff frequency.

The major concern with this model was that it failed to produce a better SNR for inputs

below 1 kHz when compared to the SNR with higher frequency inputs. Also, the SNR of the

system for any given input frequency was approximately equal to or less than the SNR produced by

a comparable Simulink model without feedback. Several small variations of this model were made,

but none of them produced more favorable results. This showed that the feedback system in its

current state was not advantageous for any reason. Because this was determined fairly late in the

project, there was no time to begin a new round of research on this type of feedback. The decision

was then made to eliminate feedback from our final design.

If full system feedback were to be implemented in the future, one of two things would need

to be done. The first option would be finding an analog to digital converter with a shorter delay

time than the one used this year. The delay time of the ADC we used was the reason such a

complicated system was needed to implement feedback. If an exceptionally fast ADC was found,

the feedback could be implemented directly, without the need for a low pass filter in the feedback

path or a second modulator. For such a model to be stable, the group delay of the ADC in the

feedback path would have to be 25 µs at absolute maximum. Even that short group delay would

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induce a phase shift of 180° at a signal frequency of 20 kHz. As was discussed in the stability

section of the background, section 2.3.1, a phase shift greater than 120° is undesirable, because it

makes the system marginally unstable.

To make our current feedback model produce better audio quality results, the transfer

function of the whole feedback system would have to be derived, examined, researched, and

improved. For our project, we only had time to examine each of the feedback system’s individual

modulator blocks, which made it difficult to assess the reason it the model was performing poorly.

More research on the topic may even reveal a better feedback model or a better way to implement

the one our team attempted.

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3.3 DSP Oriented Simulation

With a clear visual picture of the overall modulator system in MATLAB's Simulink, the next

step is to convert this model into an n-order MATLAB function with the dual purpose of simulating

the modulator similarly to the way the DSP will run its modulation code, and creating a base

program that will make it easier for us to transition to DSP code. Simulating the DSP code in

MATLAB will allow us to take advantage of MATLAB's powerful plotting and analysis tools to

evaluate and predict results in a practical way. This section describes how we implement the

modulation code in MATLAB, delves into IIR filter implementation, and shows how we took

advantage of MATLAB to optimize coefficients and predict SNR results.

3.3.1 Software Flow Diagram

The goal of our program is to generate a test signal, given a user defined input frequency and

duration, modulate the signal, given a sampling frequency and modulator order, and return an

output signal which can then be plotted and analyzed. The software flow diagram in Figure 22

details the overall operation of the modulation function DSPsim(). The MATLAB code for this

function, and all its sub-functions, is provided in appendix D.

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Figure 22 - DSPsim() Function

The user calls the DSPsim() function, providing the inputs shown in Figure 22 above. The

function has the filter coefficients for all five modulator orders predefined, and begins by using a

switch that selects the coefficients which correspond to the given order. It then creates a test signal

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that has the duration and frequency defined by the user. This signal is simply a sine wave. Then, the

modulate() function is applied for each sample of that signal. This function contains the two IIR

filters used for the modulation, uses the filter coefficients which have been previously selected, and

uses the quantize() function internally (hence the feedback path from quantize() to modulate()). The

modulate() and quantize() functions are illustrated in Figure 23 and Figure 24 respectively.

Figure 23 - modulate() Function Block Diagram

The modulate() function has the same form as that of the Simulink models. It uses two IIR

filters, called G and H, to perform the transfer function defined by the coefficients. Note that these

coefficients are passed as global variables. The output of the H filter is stored as a global variable

every time this function is called, meaning that the H value used to calculate S is that of the previous

modulation. For clarity, we can say that S[n] = input[n] - H[n-1]. The output Y is obtained by

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quantizing the result of the G filter, and then passed to the H filter. Once the H filter completes its

calculations, the new H is saved, and Y is returned, ending the function.

Figure 24 - quantize() Function Block Diagram

The quantization step is rather trivial. Threshold and output levels are defined in the

DSPsim() function and stored as global variables, and are illustrated in Figure 25 below:

Figure 25 - Tri-state Quantizer Levels

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3.3.2 IIR Filter Implementation

In many ways, the IIR function is the most important one. Its implementation and

efficiency will eventually define the maximum speed at which our DSP can modulate. While being

efficient, it must also be generalized to work for any order, so that switching between orders

becomes a simple matter of choosing the right coefficients and order. This will in turn make testing

for different orders easier. The IIR filter implementation we chose was the direct form II structure,

shown in Figure 26. It offers the advantage of being both easy to implement, and efficient.

Figure 26 - Direct Form II IIR Filter Structure (Source: MATLAB Help)

The red box shown in the figure above represents the buffer of data which contains the

output of n previous filter calculations. This buffer is then "tapped" with a coefficients to produce

the new value to store in the buffer, and then tapped with b coefficients to produce the output.

Compared to the direct form I implementation, only a single buffer is used instead of two, which is

why it is more efficient. Practically, the tapped delay line, or filter buffer, must be "circulated" every

iteration of the filter. This means that all values in the buffer need to be shifted forward to open a

spot in which the new u will be stored. The process of copying data from one array position to

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another takes a considerable amount of clock cycles, and this must be done n times per iteration,

according to the order of the filter. Therefore, minimizing buffer circulations is one way to improve

efficiency. The IIR structure in Figure 26 can be distilled into the following two equations for u (the

filter buffer/delay line) and y (the filter output).

Equation 4: Updating the current buffer line u (a taps)

Equation 5: Output : Calculating the output y (b taps)

The next step is coding the IIR filter. This process involves circulating the u buffer, then

applying Equation 4 and Equation 5 from above to calculate the new u, and return the resulting y.

Figure 27 illustrates the IIR function we created.

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Figure 27 - Software Flow Diagram for an N-Order IIR Filter Function

The dspSim() was tested and functional, and the following is a sample of plots it generated

using a 3rd order modulator, a 2kHz sine wave test signal, and a modulation frequency of 2MHz:

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0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

x 10-3

-1

0

1

Time(s)

Am

plitu

de (V

) Input test signal (2kHz sine wave)

0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

x 10-3

-2

0

2

Time (s)

Am

plitu

de (V

) Modulator output

0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

x 10-3

-1

0

1

Time(s)

Am

plitu

de (V

) Filtered Output

102

103

104

105

106

107

-100

0

100

Frequency (Hz)

Mag

nitu

de (dB

) FFT of Filtered Output

Figure 28 - dspSim() Output Sample

3.3.3 Coefficient Optimization

With a tested functional modulator simulator, we were then able to optimize the modulator

before moving on to the implementation on the DSP. In order to find a better set of coefficients

for each order, we took advantage of MATLAB's fminsearch() function to maximize SNR by

tweaking the coefficients as a parameter. Fminsearch uses an algorithm known as the Nelder-Mead

simplex method, which tweaks the function's input parameters little by little until the value returned

by the function is minimized. To take advantage of this function, we modified DSPsim() to accept a

set of coefficients as parameters rather than have them predefined, and had it calculate and return

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the SNR. The method for calculating SNR is discussed in section 4.3.1 SNR. We also made a new

function called estimateCoeffs() which takes coefficients as its only input parameter, in one matrix,

parses the coefficients, and passes them to the DSPsim() function which returns an SNR. The SNR

is negated before being returned because fminsearch finds the minimum. After being returned by

fminsearch, it is negated again to make it positive. The Matlab code for this function can be found

in Appendix E Coefficient Optimization MATLAB Code. To automate the calculation of new

coefficients for every order, we made a function called justDoIt() which uses the original coefficients

obtained from the delta-sigma toolbox as “ballpark” coefficients for each order, and passes these to

the fminsearch function to calculate a new set of coefficients for each order. The Matlab code for

this function can be found in Appendix E Coefficient Optimization MATLAB Code. Our

optimization functions were run twice. The first time, we used fewer iterations for the fminsearch

function, and this took about 2 days of calculation. The second time we increased the iterations and

let it run for a week to see if we could obtain better results. Indeed, we did see an improvement in

average SNR, as shown in Figure 29.

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Figure 29 - Average SNR Improvement from Optimized Coefficients for Orders 1-5

Note that the above plot shows the average SNR, which is the average, for each modulator

order, of all SNR's resulting from all input frequencies. Because it takes days to optimize these

coefficients, we only ran this for a modulation frequency of 2MHz. It is interesting to note how

average SNR saturates at around 88 dB starting from third order. This indicates that we can get

away with using only a third order modulator and obtain the same SNR results, while saving

processing time and having less THD than with higher order modulators. It will become apparent

in the next section how there is a compromise between SNR and THD as the order of the

modulator increases.

3.3.4 SNR Simulation Results

Using the new optimized coefficients, we ran DSPSim() for input frequencies ranging from 1

kHz to 20 kHz, covering most of the audio band. This was done for five different orders, and for a

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modulation frequency of 2MHz. Each result was passed to our THD and SNR calculation

functions to yield the plots in Figure 30 and Figure 31 for SNR and THD, respectively.

Figure 30: SNR vs. Input Frequency, Simulation

The overall trend of SNR seems to be constant as frequency increases. The average SNR

increases with higher orders. Fifth and fourth order seem to be on the same level, but fifth order

has greater variation, hence reaching lower lows but also higher highs, having a peak SNR at around

96dB. It is interesting to note how SNR spikes at 10 kHz for first and second order, and at 16 kHz

for first order. Although it is hard to see, first and second order also spike all the way up past 90dB

at 20 kHz. The reasons for these spikes are unclear.

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Figure 31: THD vs. Input Frequency, Simulation

Like SNR, % THD also increases as a function of modulator order. Higher THD is

however undesirable, so we now see that there is a compromise between THD and SNR. The order

we choose must be low enough to ensure THD is below 0.5%, and high enough to ensure SNR is

above 95 dB. Let us first consider the shape of the THD plot. There is an interesting oscillation,

where THD is high for odd multiples of 1 kHz, and low for even multiples. The smoothness of the

lines may be misleading, because we really only have points every 1000 Hz. If we had more points,

it is likely that we would see higher frequency oscillation added to the current oscillation. The

overall trend of the THD also seems to be oscillating, as the peaks start low at 5 kHz, increase until

they reach a peak at 13 kHz, and begin to decrease. If we had more data past 18kHz we may have

seen the peaks continue to decrease, then begin increasing, following the general oscillating trend. If

we were to write a formula to recreate this plot, it would be a sum of sine waves that range from low

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to high frequencies, where the amplitude of these waves is increased as a function of order. THD is

calculated as the power of the original signal's frequency over the sum of the powers of its

harmonics. Our THD plot indicates that as input frequency shifts from left to right, the original

frequency power will decrease, while the sum of its harmonics' powers will increase, then the

opposite will happen. What causes this behavior, however, remains unclear. The conclusion we

can draw from this much speculative reasoning is that higher orders will create more radical

oscillations by increasing the amplitude of all sine waves which add up to the resulting THD plot for

each order in figure 35.

With respect to our desired results, if we were to take the average of all points for each

order, the highest %THD will be around 0.6% for fifth order, 0.25% for third and fourth order, and

0.1% for first and second order. According to our two plots, the best choice is fourth order because

it has higher SNR than third order, and only slightly higher THD than third order.

3.4 DSP Modulator Implementation

With a working DSP simulation, we were ready to implement the modulator on the DSP.

However, new dimensions of complexity needed to be taken into consideration due to the nature of

DSP's. This includes interrupt timing, sampling restrictions, core clock speed consideration, core

hang issues, and overflow problems. A solution was found to solve each of these problems, all of

which are described in this section, along with the overall system flow of our DSP program.

3.4.1 System Flow Diagram

The main difference between the DSP code and MATLAB code is that, instead of having a

predefined input in a nice buffer of data, we are now sampling the input in real time. This means

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that we need to use timed interrupts to obtain the samples from the audio codec. The overall

system diagram is simplified in Figure 32, and detailed in Figure 33.

Figure 32 - Simplified System Flow Diagram

Figure 32 above is color-coded such that each colored block matches its corresponding

timing diagrams shown in Figure 34 and Figure 35.

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Figure 33 - Detailed System Flow Diagram

The code running in each timer interrupt is almost identical to that used in our Matlab simulation.

The main differences are:

• We now need to setup the hardware to read from the proper input ports and write to the

desired output pins.

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• The SPDIF interrupt is used to acquire data, saves this data, but does not call the timer

interrupt.

• Each time the timer interrupt is triggered, it uses whatever data is available in the “save input

sample” block shown in Figure 33.

The code used in the DSP is provided in Appendix G DSP Code.

3.4.2 Timing

Having a clear picture of how the timing of the input sampling relates to the timing of the

core clock, and the timer interrupt, is crucial to understanding how and why the DSP program

works. Figure 34 shows a sample timing scenario for a sampling frequency of 96 kHz, a core clock

running at 397MHz, and a timer interrupt service routine running at the modulation frequency, or

2MHz. One can see how the modulation, which occurs once every timer ISR iteration, occurs

multiple times per sample. Each time the modulation function is run (filter + quantize), the existing

input sample is used, implying that we are using zero-order hold to output at a frequency of 2MHz.

The reuse of input samples is acceptable in our case, because doing so will not lower our SNR below

our goal.

Figure 34 - System Timing Diagram

Figure 35 shows the timing dependencies and non-dependencies between the various signals

and functions. What’s important to note is that the modulation is independent from the sampling

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frequency of the SPDIF interrupt. The SPDIF ISR actually requires very few clock cycles to execute

because its only function is to acquire the input sample from a register, convert it to floating point,

and store it as a global variable. We have setup the two interrupts such that the SPDIF has a higher

priority than the timer one. This is important because we do not want to drop any incoming

samples. Once every sampling interval, the SPDIF will break the operation of the timer ISR, and

update the input sample. This does not negatively affect the modulation because:

• The filter stage reads the input sample once at the beginning of its function, and does not

use it again.

• The SPDIF interrupt duration is shorter than the wait time shown in Figure 35.

Figure 35 - Timing Relationships

As a note to avoid confusion, the timing shown for the two ISR’s in the figure above does

not represent the execution time of each ISR. Rather, each complete cycle represents the time

between two subsequent iterations of the same ISR. This also applies to the timing in Figure 34.

Furthermore, the time measures shown in the quantize line in Figure 35 are all variable. That is, the

time to execute the G filter for one sample may not be the same time for the next sample. For this

reason, and to account for instances where time is needed to perform the SPDIF ISR, we have a

wait time that will vary from one modulation cycle to the next. The digital output pins are only

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updated when the next timer interrupt occurs, to make sure the outputs are equally spaced in time,

and to account for randomness in modulation time.

3.4.3 Interrupts and their Pitfalls

This design calls for a significant number of operations in order to generate a single

modulator output. As discussed in the previous sections, the DSP must perform multiple IIR filter

operations, quantization's, and sample receiving while maintaining accurate timing between

interrupts, and continued operation even at high output frequencies.

While interrupts make this type of design possible, they also add extra complications that are

impossible to traditionally debug. When user interrupts are being nested, ensuring proper timing

becomes difficult. The number of cycles required to initiate and return from an interrupt depends

on the language and the processor being used. The number of interrupts used for this project may

not be functional on other processors if the interrupt requires too many additional cycles. The

particular processor used from this design had a range of interrupt types with varying cycle

overhead. The fastest interrupt was not usable for the design, because some important registers

were not preserved. If this design is moved to a new platform care must be taken to ensure the

proper interrupt type is used in order to preserve the exact timing diagram as shown in the previous

section. Errors in the interrupt order could cause dropped samples, or extended missed modulator

outputs, both of which could cause audio degradation.

Not all interrupts are treated equally by the DSP. Certain system interrupts occur with a

higher priority than user interrupts. With outputs in the MHz range any system interrupt, even short

ones, can cause a long modulation pause. In the particular processor used for this design, the analog

to digital converter interrupts had an odd behavior. The analog to digital converter uses multiple

buffers. The basic operation is that the converter uses a register for the latest sample that is being

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received and another register for the sample that was previously received. After the previously

received buffer is read, it is then rotated with the register that is used for the current sample. When

the analog to digital converter generates it's interrupt the core hangs until the receive buffer is read.

If the receive buffer is not read the process core will hang temporarily. This causes a pause in the

modulator output, but the modulator will continue to run. This is the type of error that was

common during debugging. Because the modulator continued to run, and the audio output

continued to work, it was not immediately obvious that there was an output error until the output

was buffer and analyzed.

3.5 Hardware and PCB Design

The design of a new PCB was vital to the implementation of full system feedback. Since this

project reused the power stage and filter design from last year, much of the PCB layout was able to

be reused. The PCB for this year’s project was redesigned with feedback and modular connections

in mind.

The initial design of the feedback network was overwhelmingly simple. The design simply

added a capacitor and resistor series to each side of the output speaker, creating a DC blocked and

attenuated version of the output. One half of the output signal was fed to the left channel, and the

other to the right channel of the DSP's analog to digital converters. The maximum output of the

amplifier was estimated as being equal to the voltage applied to the power stage. The output was

attenuated to use the full scale range of the ADC for each channel. Within the programming of the

DSP the two channels were subtracted, creating the attenuated output.

This setup had several problems. This method required the use of two ADC channels, and

our final DSP kit only had two channels. This means we were unable to accept an analog input

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when feedback was implemented. This method was also lacking in its ability to reject common

mode interference. The power stage of the Class D amplifier exhibits EMI that could distort the

relatively low voltage feedback signal. The ADC's of the DSP are not designed to reject common

mode noise, especially from such a close and powerful EMI source.

A more suitable method for feedback signal capturing utilized a differential to single ended

converter built from a differencing op amp. This method still utilized a capacitor and resistor in a

series combination to DC block and attenuate the single, which ensured that the output was within a

range that an op amp with a +-12V supply could handle without clipping the input.

This was the method that was chosen, although a differencing op amp was not used.

Instead a differential to single ended converter, THAT1200, designed to be an ADC front end with

high common mode rejection and built in attention, was used.

Figure 36 - Feedback Capturing Network

The PCB was designed with common mode rejection in mind. Because the DSP was used

to both generate the input modulation signal and receive the feedback signal, both these I/O's were

placed on the same side of the board. This meant that the feedback signal would have to travel the

entire length of the board and cross a high power copper plane split. The signal was kept

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differential until the very end of the board, ensuring that the ADC driver could reject the noise

added by the power stage.

Figure 37 - Class D Audio 2009 PCB

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4 Testing

To ensure that the design met its goals, a series of tests were developed to acquire a wealth

of data points from the system. Testing was performed using a mix of hardware, such as scopes and

precision power supplies; and software, like MATLAB. Hardware testing was straightforward, and

did not involve any elaborate setups. However, software testing, in MATLAB, required complicated

code.

There were three main categories for testing. The first category was power and efficiency.

This was easy to test as it only required the use of some basic lab equipment, careful observation of

the measurements, and some simple math. The second category was input testing. This was even

easier to test as it only required finding appropriate test sources to provide digital or analog inputs.

The third was sound quality. Testing sound quality required audio capturing as well as software

based computation of the signals.

Figure 38 - Test Bench (Left to Right: Resistor Test board, Amplifier PCB, DSP)

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4.1 Power and Efficiency

The setup for testing the power output of the system was simple. The output power is

calculated using the output amplitude, measured at the speaker, as well as the resistance of the

speaker itself. Because testing lasted a long time, a series of resistors, totaling 8-ohms, was used as

the output load as opposed to a speaker. Placing a speaker on the output of this 80+ Watt amplifier

would have been very disruptive when running at full power. The exact resistance of the speaker

was measured using a precision Hewlett Packard 3458A multimeter. The output, RMS voltage, was

measured using a Tektronix TDS784C oscilloscope.

Testing for efficiency was slightly more involved. The efficiency of our system was

determined by the ratio of its input and output power. The PCB for the system called for two

power rails. The first was a low voltage, low current rail required to power the gate drivers. The

gate drivers were isolated from the actual amplification process because they only made contact with

the near infinite input resistance of the MOSFET gates; as a result their input power did not

fluctuate greatly.

The second rail powered the amplifier's MOSFETs. At any given time, current delivered to

this rail will have passed through two MOSFETs, two inductors, and the load (speaker). The power

delivered by this rail was the overwhelming contributor to the input power calculation. The voltage

applied to the rail directly affected what output power is achievable.

The rail for the gate driver was powered by a common Tektronix PS2521G programmable

power supply, as its requirements were not out of the ordinary. This supply offered a two digit

current display which was used to calculate the input power for the gate drivers. The rail for the

MOSFETs required a high power supply so the BK Precision High Current DC Regulated power

supply was used. This power supply only had a single digit current display, so an in-line Hewlett

Packard E2373A multimeter was used to measure the supplied current.

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4.2 Input Sources

One of the goals for this design was to be able to support both analog and digital inputs, so

both of these needed to be tested. The analog input for the system was tested using a function

generator. The input was tested for a series of frequencies and amplitudes, ranging the width of the

audio band and the peak input of the ADC. Subjective listening tests were also performed with

devices like PC's and iPods. The digital input was tested using a DVD player with an SPDIF output.

A CD was generated that contained a similar range of frequencies as the analog test, but it was

limited to a single amplitude. Subjective listening tests were also performed using the same DVD

player and music CD's.

4.3 Audio Quality

Testing audio quality was much more challenging than the previous tests. Audio quality

measurements include SNR, zero-input SNR, and THD. Measuring these involved recording the

output of the amplifier for various inputs and determining the SNRs and THD of these

measurements.

The audio was captured using a 192 kHz 24-bit sound card, the EMU-0202. This sound

card had a limited input range of approximately 2.7 Vpp, so the output needed to be attenuated

before recording, as the input of the sound card would be constantly peaked otherwise. The

attenuation was performed by the feedback driver built onto the PCB. While feedback was never

implemented in the DSP, the hardware portion was built, and this served to both attenuate the

output as well as convert it from a differential to a single ended signal. The audio signals were

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recorded into SoundForge6, a piece of software capable of recording with a sampling rate of 192

kHz.

The test input signals for the system were generated within the DSP. The primary input of

the DSP is a digital source, which experiences no distortion due to noise, so there is no difference

between inputting a sine wave from a digital source, and generating the sine wave within the DSP

during the digital signal buffer's receive interrupt. This allowed greater control of the test frequency

pattern.

The testing process was automated within the DSP. After loading a specific set of

modulator coefficients, and settings the desired modulation frequency for the modulator, the DSP

would begin to output 1 second clips of 1 kHz spaced frequencies. After ranging the input from

20Hz to 20 kHz, the order of the modulator was increased. This allowed the testing of nearly 100

different input frequency and modulator order combinations in just under a minute. Recordings

were captured for three different modulation frequencies, 5 different orders per modulation

frequency, and 15 frequencies per order. After the recordings were taken they were sent to

MATLAB for processing.

4.3.1 SNR

In MATLAB, several functions were written to find the zero input SNR, SNR, and THD of

an audio clip and to automate the process of calculating these statistics for all of the different data

that we had collected. First, the method used for calculating both SNRs will be examined. As was

discussed in section 2.2.3, SNR is defined as in Equation 6 below.

Equation 6 - Definition of SNR

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There are two different ways that the power of the noise can be determined. For zero input

SNR, the power of the noise is found by recording the output of the system when the input is zero.

For SNR, the noise is anything in the recorded output that is not the desired signal. We chose to

calculate both of these parameters in order to compare them to each other and the results of other

Class D MQP projects. First, our method of calculating zero input SNR will be examined.

The noise of the system for the zero input SNR was taken as a recording of the amplifier’s

output while the input to the amplifier was a mathematical zero. The noise power was calculated

directly from this recording. To find the signal power for this case, separate recordings for various

different input frequencies were acquired. The power of the desired signal in these recordings was

taken as its maximum dB magnitude value in the frequency domain. After calculating these two

parameters, the formula from Equation 6 above was used to determine the SNR. The MATLAB

code used to calculate the zero input SNR can be seen in appendix F. Although the zero input SNR

gives us a good metric for comparison to previous MQP projects, the process of determining a more

realistic SNR will be examined next.

In order to calculate the SNR, both the power of the signal and the power of the noise must

be calculated again. The problem is that both the noise and the desired signal are combined in the

output. Luckily, we know that the desired signal, or what the output would look like if the system

added no noise at all, is a pure sine wave. This fact allows us to separate both the noise and the

desired signal from the recorded output. Because the amplifier is a linear system, the desired output

sine wave must only be a shifted and amplified version of the pure input sine wave. The general idea

is to create the desired signal from the output, use that to find the signal power, and subtract it from

the recorded output to find the noise and hence the noise power.

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In order to create the desired signal sine wave, we had to create an amplified, time delayed

version of the input signal. Shown below is an equation for this sine wave, with  as the

amplification factor, ố as the delay factor, y as the desired output, and cos(t) as the input sinusoid.

. � Â � cos "3 & ố$ Equation 7 - Desired Sine Wave Output

The values of  and ố for any given signal can be estimated as shown in Equation 8 and

Equation 9 below, respectively (Kay, 1993). In these equations, y’ is the given signal, f is the

frequency of the signal, n is the number of samples in the wave, and Fs is the sampling frequency of

the wave.

 � 4.5 � 6�7��8�9�:� 4

; � 2

Equation 8 - Formula for Estimating Amplification Factor Â

ố � tan��@A.5 � sin C2 � D � E � ;F# GH

.5 � cos "2 � D � E � ;F# $

Equation 9 - Formula for Estimating Time Delay Factor ố

After finding the values of  and ố, the desired output can be built according to Equation 7

above. Calculating the power of this desired output yields the signal power. Subtracting the built

signal from the actual output yields the noise signal. The noise power is calculated from that signal.

Once the power of the desired signal and the noise is known, these values are plugged into the

formula for SNR from Equation 6. The actual MATLAB code used to implement this method can

be found in appendix F.

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4.3.2 THD

Total harmonic distortion is defined as the ratio of the power of a signal's fundamental

frequency to the power of its harmonics, expressed as a percentage. To obtain results that are

comparable to similar Class D projects from previous years, we will only measure the first five

harmonics. If we let Pf be the power of the fundamental frequency, and Pn be the power of the nth

harmonic of that signal, then THD is calculated as follows:

%JKL � M9NM� & M & MO & MP & MQ

� 100

Equation 10: THD formula

To obtain the power of the fundamental frequency and its harmonics, the first step is to take the

FFT of that signal. The power of any frequency within that FFT is simply its magnitude. In Matlab,

we obtain the power of the fundamental frequency by finding the max value in the FFT. The index

of this value is then multiplied by n to give us the index of the nth harmonic, which we then use to

retrieve its value from the FFT array. We do this for the first five harmonics, square them, and take

the square root of their sum, by which we divide the power of the fundamental frequency, and

multiply the dividend by 100 to yield the final % THD. For the Matlab code used to calculate THD,

refer to appendix F (getStats.m).

4.3.3 Automating Sound Quality Testing

To obtain an adequate amount of data for analysis, while minimizing manual labor to obtain

it, we automated both the generation and acquisition of a range of output data from the DSP, the

measurement of the results, the formatting of the results, and the plotting. First let us consider the

variables we are dealing with. We want to test SNR, THD, and efficiency for three different

modulation frequencies: 1MHz, 1.5MHz, and 2MHz. We chose these values because our DSP

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cannot modulate faster than 2MHz due to timing restrictions, and 1MHz is already below 60x

oversampling requirement. For each of these, we need to test five different modulator orders. Fifth

order is the maximum our DSP can support. A higher order would require more time than allowed

by the slowest timer ISR at 1MHz. For each order, we want to test enough frequencies to generate a

meaningful plot. We chose to test for nineteen different frequencies, equally spaced along most of

the audio band, from 20Hz to 19.20 kHz. We now have three dimensions: modulation frequency,

order, and frequency. The total number of SNR calculations needed is 3*5*19 = 285. The same

goes for THD.

Automating the Generation of Test Signals:

Instead of actually sampling an SPDIF input for 285 different possibilities, we decided to

generate each within the DSP. This does not compromise the credibility of the results because an

SPDIF signal is a digital signal not unlike the digital signal we generate in the DSP. To stay true to

the interrupt restrictions, we simulate sampling by incrementing a time variable by 1/96 kHz every

time the SPDIF interrupt is called, and take the sine of this value times the desired frequency to

produce a sample, as seen from the following DSP code extracted from the sampling interrupt:

timeADC+=time_diff; //time_diff = 1/96000 freqNum = timeADC/0.5; //increment frequency inde x every 0.5 seconds adc = cos(2*PI*freq[freqNum]*timeADC); //generate sample

In the main function, we create an array freq[] than contains 20 different frequencies. These

frequencies start at 20Hz, and increase by 1000 every iteration until they reach 18.20 kHz, for a total

of 19 different frequencies. In the sampling ISR, we fetch a new frequency from freq[] every 0.5

seconds. In the timer ISR, we check if freqNum, the frequency counter, has passed 19, in which

case we reset it to 0 and increment the modulator order. As a result, if we let our program run, it

will modulate and output 19 frequencies per order, with a duration of 0.5 seconds per frequency, for

5 different orders. Our manual task is to change modulation frequency twice. With this, we were

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able to easily let the DSP play and record the filtered output, then split up the recording into

multiple wave files for each frequency.

Automating the Measurement of the Test Signals:

Having generated and recorded all the necessary test signals, we now have to measure the

SNR and THD of 285 different wave files. This process had to be automated as well. The first step

was naming the wave files such that we could easily parse the file names to determine its order,

modulation frequency, and signal frequency. We then wrote a function called getStats(), which uses

the following logic to calculate the THD and SNR of each of these wave files:

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Figure 39: Flow Chart for Automating SNR and THD Calculations

The overall idea is to have three levels of loops, j for the three different modulation

frequencies, i for the five different orders, and n for the nineteen different frequencies. Within each

iteration, the function readFile() will be given the current value for i, j, and n, which will let it

determine which file to read and return. The readFile() function will use the given i, j, and n to

generate the name of the file according to the naming convention we used, then save it in an array

called signal and return it. This array is then passed to a function to calculate SNR, and then a

function to calculate THD. When SNR is calculated, the result is saved in a three dimensional array,

with the indices being the current i, j, and n. The same goes for THD. When all loops have

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terminated, the function returns these two arrays. The code for getStats() and all the functions it

calls is found in Appendix F.

We must note that a slightly different version of the getStats() function, called

getZeroSNR(), was used to measure the SNR of zero-input measurements, using the zero-input

SNR calculation method. This introduced a fourth dimension to our array of results, because in

addition to 3 modulation frequencies, 5 orders, and 19 frequencies, we now had 3 different types of

zero input measurements. The first type was the measurement of noise with the modulator powered

on but with no input given to it. The second type was the noise measured with the modulator

powered off. Finally, we measured the noise with only the feedback attenuation circuit powered on.

The difference between getStats() and getZeroSNR() was the SNR calculation function used, an

additional file being returned from the readFile() function, called noise, and a fourth loop to

encompass the 3 different types of zero input. The results of getZeroSNR() did not include THD,

and was a four dimensional array of SNR measurements. Its code is also found in Appendix F.

Automating the Parsing and Plotting of Measured THD's and SNR's

We now had three-dimensional matrices for both THD and SNR, and a four-dimensional

matrix for zero-input SNR. This made the task of plotting the results rather complicated. We

therefore wrote a function to reformat these matrices into structs that included 15 by 19 arrays for

THD, SNR, and frequency. The 19 columns represented the 19 different frequencies, the first 5

rows were orders 1 through 5 for a modulation frequency of 1 MHz, the next 5 rows for 1.5 MHz,

and the final 5 rows for 2 MHz. The function used to format was called format() and is found in

Appendix F under format.m. Finally, we wrote a function called plotForUs(), which took any two

formatted results, and plotted one against the other on a single plot, labeling the axes and title

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accordingly, adding a legend, and color coding the results for the five different orders. This allowed

us to plot, using a single function call, frequency vs. SNR, frequency vs. THD, or even SNR vs.

THD, although the latter did not prove to be of any use. The code for this function is found under

plotForUs.m in Appendix F.

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5 Results

The goal of this project was to build an 80 Watt amplifier with both a digital and analog

input that could achieve over 95% power efficiency, more than 100 dB SNR, and less than 0.5%

total harmonic distortion. Our final amplifier design met every aspect of this goal, making the

project a success. In this section, the detailed specifications of the amplifier will be examined. The specifications do not explain how we met one of our goals, to build an amplifier that

had both digital and analog inputs. However, this goal was met and tested thoroughly. After the

amplifier design was finalized, a DVD played with an SPDIF (Digital) output was used as the input

to our amplifier. The final test for this functionality was run by playing an audio CD in that DVD

player and listening to the output of the amplifier by attaching a speaker to the load. The music that

played was qualitatively "good" sounding. The final test of the analog input was done using the

same DVD player's analog output, which produced an output with the same level of quality as the

digital input. The data used to calculate the SNR, THD and efficiency power results that follow was

collected while a digital signal from inside the DSP was used as the input to the modulator.

5.1 SNR

As was discussed in the testing section of the report, there are actually two methods of

calculating the signal to noise ratio of the amplifier for any given input. Initially, our SNR goal of

100 dB was written with the intention of improving upon previous Class D audio amplifier MQPs.

Because the official SNR used for last year's Class D project was zero input SNR, we calculated the

same parameter for a comparison. The results of this test were even better than achieved in

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previous years. As can be seen in Figure 40, Figure 41, and Figure 42 below, the average zero input

SNR of our modulator was roughly 109 dB for all orders and modulation frequencies. The zero

input SNR does drop significantly at an input frequency of 18 kHz for every modulation frequency.

The raw data used to create the zero input SNR figures below can be found in appendix H.

Figure 40 - Zero Input SNR vs. Input Frequency for 1 MHz Modulation Frequency

Figure 41 - Zero Input SNR vs. Input Frequency for 1.5 MHz Modulation Frequency

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Figure 42 - Zero Input SNR vs. Input Frequency for 2 MHz Modulation Frequency

These zero input SNR numbers easily accomplish our goals, but do not make much sense in

terms of the qualitative definition of signal to noise ratio. Because our amplifier is driven by a digital

signals processor, the amplifier can effectively be turned off every time the input is zero. This fact

made our zero input SNR values somewhat non-descriptive of the actual performance of our

system. In order to more effectively measure the noise reduction of our amplifier, the actual SNR of

several different output samples were calculated. This data is displayed below in Figure 43, Figure

44, and Figure 45. There are two main trends in these figures. The first and more obvious one is

that SNR is steadily decreased with increasing frequency. This makes sense, because as the input

frequency goes up, the oversampling ratio of the delta-sigma modulator goes down and is therefore

less effective at approximating the input. This steady decline is impressive, because each of the plots

contains data from 18 different input frequencies. The second trend is the fact that higher order

modulators seem to produce worse SNR results, with the exception of first order. The fifth order

modulator produced the overall worst SNR results, which is counter intuitive. The first order

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modulator also performed very poorly, which was expected. The raw data used to create the figures

below can be found in Appendix I.

Figure 43 - SNR vs. Input Frequency for 1 MHz Modulation Frequency

Figure 44 - SNR vs. Input Frequency for 1.5 MHz Modulation Frequency

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Figure 45 - SNR vs. Input Frequency for 2 MHz Modulation Frequency

5.2 THD

The results of the THD testing also yielded numbers below our goal of 0.5%. As can be

seen below in Figure 46, Figure 47, and Figure 48 below, the THD numbers for all five orders are

almost always under 0.5%. There are exceptions, such as certain input frequencies when the

amplifier was running at 1 and 1.5 MHz modulation frequencies and using first order coefficients.

At a modulation frequency of 2 MHz, however, this pattern seems to break and it is the fifth order

modulator which yields the worst THD numbers, followed by forth order. The fifth order

modulator even produces THDs above 1% for a few input frequencies, which is much higher than

seen under any other circumstances. The raw data collected to create the figures shown below can

be found in Appendix J.

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Figure 46 - THD vs. Input Frequency for 1 MHz Modulation Frequency

Figure 47 - THD vs. Input Frequency for 1.5 MHz Modulation Frequency

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Figure 48 - THD vs. Input Frequency for 2 MHz Modulation Frequency

5.3 Efficiency and Power

Efficiency and power are mostly dependant on the power stage and filter, but there is some

influence from the modulator scheme. In order to determine the influence that a modulator has on

the efficiency and power output of a Class D amplifier, test points were accumulated for multiple

frequencies, with multiple modulator orders and modulation frequencies. The raw power and

efficiency data that was collected can be seen in appendix K.

The power output goal of the project was clearly met, reaching values above 90 watts. The

figure below shows the output power versus the input frequency for two different modulator orders.

In both cases the modulator frequency was locked at 2 MHz, and the input was locked at an

amplitude of 2. Since the input is digital, the input amplitude is rather vague and the importance of

the input amplitude is the relative size. And input amplitude of 2 is the largest input that was used,

larger values become unstable for many modulator orders.

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Figure 49 - Output Power vs. Frequency

(2MHz modulation frequency, high input amplitude)

Some of the higher order modulators became unstable at higher amplitudes. In order to get

a relative scale of power output versus modulator order, another data set was created using a lower

input amplitude. This resulted in lower output power, but gave better insight into the differences

between the orders. From the following plot two possibilities result. The first is that the test data

for the 1st order modulator was somehow skewed, resulting in a very low reading. The other

possibility is that there is an exponential relationship between modulator order and output power

which is asymptotic to a certain value.

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Figure 50 - Output Power vs. Frequency

(2MHz modulation frequency, medium input amplitude)

The results for efficiency proved to be equally interesting. The figure below shows the

average efficiency for each modulator, with an input amplitude of 1.5 and a modulation frequency of

2 MHz. The first three orders show the efficiency of the system asymptotically approaching 95.5%

by the third order. However; the 4th order efficiency results take a sharp dive to below 94%

efficiency.

Figure 51 - Average Efficiency vs. Modulator Order (Input amplitude 1.5, 2MHz modulation frequency)

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The degree to which the 4th order efficiency drops appears to be a result of some outlier

data points, namely 20 kHz. For some orders the efficiency at 20 kHz was very similar to the

efficiency of the other frequencies. In the figure below the, efficiency for only the 20 kHz data

point for each order is shown.

Figure 52 - Efficiency for All Modulator Orders at 20 kHz

It is clear from this figure that the average efficiency is dominated by this point. There also

appears to be some correlation between the 1st and 4th orders with regard to the efficiency of the

system. The following figure is a plot of the average efficiency of the different modulator orders

with the 20 kHz data point removed from each order. While the 4th order modulator still adds

lower average efficiency, it is now above 94.5%.

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Figure 53 - Average efficiency vs. Modulator Order

With the removal of both 20 kHz and 12 kHz the 4th order modulator becomes one of the

most efficient orders that was tested. The following plot shows the efficiency for only the 4th order.

The plot shows consistently high efficiency until 5 kHz.

Figure 54 - Efficiency of a 4th Order Modulator

This final efficiency plot is the only one that has been displayed that has used an input

amplitude of 2, previously described as a large input value. As seen from the power plots, higher

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input amplitudes result in higher output power. This also corresponds to higher power efficiency.

The following plot shows efficiencies consistently above 97%.

Figure 55 - Efficiency vs. Frequency

6 Conclusion

The initial goal of this MQP was to create a Class D amplifier that incorporated a digital

input, full system feedback, over 100dB SNR and less than 0.5% THD. These goals required

significant time and resources which would have made the task of designing an entirely new system

too lengthy to complete in the normal MQP time frame. The previous year's Class D team created a

solid power and filter design which we were able to reuse for our new design. Since the output

power and efficiency are greatly dictated by the power stage our goals for these specifications were

based off the results of the previous year.

Modulator design progressed mostly inside MATLAB. Through countless simulations the

modulator was tweaked to create as high quality of an output as possible. Within MATLAB a virtual

DSP was generated to aid in the testing of new modulator designs, and ease the transition from

theoretical to practical. While MATLAB simulations were progressing the DSP was studied and its

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functionality explored in order to combine the simulation with the real device as seamlessly as

possible.

The incorporation of feedback was greatly explored with in simulation. Many different

configurations were tried, but no configuration was capable of producing any quality gains. While

feedback simulations progress, the circuit required to capture the output and deliver it to the DSP

was created. This circuit was designed onto a PCB along with the previous year’s power and filter

stage designs.

The DSP approach to the Class D amplifier proved to have many benefits over traditional

configurations. In previous years teams have been forced to generate simulations, work out as many

flaws as they can, decide on the best configuration of OSR and modulator order, and then construct

their design in hope that there would be no errors. The DSP allowed many different modulator

configurations to be applied to the same power stage and filter, allowing real life testing and

comparison of limitless modulator configurations.

The DSP also aided in the successful implementation of a dual input system, allowing the

amplifier to work with both analog and digital signals. With this configuration the design was able

to amplify audio from simple devices like portable CD players, and more complicated devices like

DVD players with coaxial digital audio connections.

The project also resulted in a catalog of useful MATLAB tools. These tools aided in the

automated testing of many data points to determine SNR and THD numbers as quickly and

accurately as possible, tools which could be reused in future years. It also resulted in code designed

to automate the discovery of new and theoretical optimized coefficients for use in discrete time

modulators.

This project helped redefine the audio quality requirements for the digital input Class D

amplifier. Through careful examination it was concluded that traditional audio quality testing

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techniques, used for determining SNR, cannot be accurately applied to a digital input Class D

modulator. As a result, MATLAB tools used to calculate SNR were designed to be robust in their

ability to determine audio quality for specific inputs.

The final product of this MQP was a Class D audio amplifier that produced over 90 watts of

output power, and was over 95% efficient. It was capable of over 110dB of 'zero-input' SNR, and

over 50dB of standard SNR accepting both analog and SPDIF inputs.

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7 Future Work and Recommendations

The Class D amplifier project runs nearly every year at WPI. Most years that the amplifier is

developed the final report outlines recommendations to future Class D amplifier teams. One such

recommendation was the incorporation of a digital input, which was implemented this year. It has

also been suggested that full system feedback be added, which was attempted this year although it

was not successful.

One of the most obvious, and commonly recommended additions is a high efficiency power

supply for the Class D amplifier. The Class D amplifier requires a ripple free, high power DC

supply. The cleanliness of the power supply rail directly affects the quality of the output. While a

Class D amplifier is efficient, the conversion from AC to DC can be an extremely inefficient,

negating the selling points of the Class D. Because the Class D amplifier requires a DC power

supply, it has a perfect application in automotive audio. While an AC to DC power supply may be

good, and DC to DC power supply capable of conditioning automotive power, which can fluctuate

between 11 and 16 volts, would be a very good addition to the project.

With this year’s addition of a digital input, the amplifier could be expanded to support

multiple channels. DSP's have ample digital output pins that can be used to drive multiple power

stages. SPDIF is capable of transmitting more than one channel. These channels could be extracted,

separately modulated, and sent to their respective power stage.

It is common for consumer amplifiers to have signal processing features that provide the

user with control over their listening experiences. Now that the Class D amplifier has been built

into a DSP, a user interface could also be included to allow the user to adjust the EQ settings of the

amplifier in order to compensate for low quality speakers or abnormal room conditions. Assuming

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the amplifier was servicing multiple channels it could be programmed to send high frequency to

tweeters while sending low frequencies to mids or sub-woofers.

The output filter for the Class D amplifier requires the speaker to be the one resistor in the

circuit. This has a few implications for the design. Variations in speakers, even among speakers

rated as 8 ohm, will result in variations in their resistance. This will cause the filter to change cutoff

frequency. The 8 ohm speaker is not the only available type. The 4 ohm and 6 ohm speakers are

also common; these would drastically affect the filter cutoff. Speakers do not act as perfect passive

devices. A speaker has a large coil and magnet used for pushing the cone. The speaker and magnet

can have rippling affects that travel back through the amplifier and cause distortion. A good future

addition would be the ability to support multiple speaker types, and incorporate circuit protection

allowing the device to remain powered on even with no load attached, which could currently cause

damage to the MOSFETS.

The Class D amplifier does not have to be relegated to the audio market. Class D amplifiers

can be developed with a band-pass or high pass modulator, as opposed to a low pass modulator.

The Class D amplifier can offer high efficiency amplification to devices in the RF region. A Class D

amplifier could be configured as a band pass filter in the RF band and used to efficiently amplify

digital and analog signals for communication or any long range wireless transmissions. It would be

especially helpful in handheld devices which need to run on batteries.

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References

1 Dorf, R. C. (1989). Modern Control Systems. Reading, Massachusetts: Addison-Welsley.

2 Kay, S. M. (1993). Fundamentals of Statistical Signal Processing Estimation Theory Volume I. Upper Saddle

River, New Jersey: Prentice-Hall Inc.

3 Lathi, B. (2005). Signals and Systems. New York, New York: Oxford University Press.

4 Morey, B., Vasudevan, R., & Woloschin, I. (2008). Class D Audio Amplifier. Worcester, MA: WPI.

5 Schreier, R., & Temes, G. C. (2005). Understanding Delta-Sigma Data Converters. Hoboken, New

Jersey: John Wiley & Sons, Inc.

6 Tomarakos, J. (2002, July 16). The Relationship of Dynamic Range to Data Word Size in Digital Audio

Processing. Retrieved September 15, 2008, from

http://www.audiodesignline.com/showarticle.jhtml?article=192200610

7 Wikipedia. (2008, September 2). Retrieved Spetember 8, 2008, from

http://en.wikipedia.org/wiki/Pulse_density_modulation

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Appendix A CIFB Modulator Coefficient

Generation MATLAB Code

% In order to run this MATLAB script, the delta sig ma toolbox of MATLAB % must be installed % This script was derived from the code fragment on pg. 285 of % "Understanding Delta-Sigma Data Converters" by Ri chard Scgreier and Gabor % C. Temes function [a, g, b, c] = CalcCoefficients(order, OSR ); H = synthesizeNTF(order,OSR,0); form = 'CIFB'; [a,g,b,c] = realizeNTF(H, form); b(2:end) = 0; ABCD = stuffABCD(a,g,b,c,form); [ABCDs umax] = scaleABCD(ABCD); [a, g, b, c] = mapABCD(ABCDs,form)

Appendix B Loop Filter Coefficient Generation

MATLAB Code

function [NTFtop, NTFbottom, Gbottom, Gtop, Hbottom , Htop] = SynthesizePlus(order) [NTF, z, p] = MysynthesizeNTF(order); NTF.variable ='z^-1'; NTF = tf(NTF); G = 1/NTF; H = 1 - NTF; NTFtop = poly(z); NTFbottom = poly(p); Gtop = NTFbottom; Gbottom = NTFtop; Htop = NTFbottom - NTFtop; Hbottom = Gtop; function[ntf, z, p] = MysynthesizeNTF(order,osr,opt ,H_inf,f0) %This function is a modified version of synthesizeN TF, which is included in %the Delta-Sigma toolbox. Many thanks to the create r of the toolbox, %Richard Schreier. %ntf = synthesizeNTF(order=3,osr=64,opt=0,H_inf=1.5 ,f0=0) %Synthesize a noise transfer function for a delta-s igma modulator. % order = order of the modulator

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% osr = oversampling ratio % opt = flag for optimized zeros % 0 -> not optimized, % 1 -> pre-computed optima (good for high osr) % 2 -> as above with at least one zero at band-cent er % 3 -> optimized zeros (Requires MATLAB6 and Optimi zation Toolbox) % [] -> zero locations in complex form % H_inf = maximum NTF gain % f0 = center frequency (1->fs) % %ntf is a zpk object containing the zeros and poles of the NTF. See zpk.m % % See also % clans() "Closed-loop analysis of noise-shaper." A n alternative % method for selecting NTFs based on the 1-norm of the % impulse response of the NTF % % synthesizeChebyshevNTF() Select a type-2 highpass Chebyshev NTF. % This function does a better job than synthesizeNT F when order % is high and H_inf is low. % Handle the input arguments parameters = { 'order' 'osr' 'opt' 'H_inf' 'f0'}; defaults = { 3 64 0 1.5 0 }; for arg_i=1:length(defaults) parameter = char(parameters(arg_i)); if arg_i>nargin | ( eval(['isnumeric(' parameter ') ']) eval([ 'any(isnan(' parameter ')) | isempty(' parameter ') ']) ) eval([parameter '=defaults{arg_i};']) end end if f0 > 0.5 fprintf(1, 'Error. f0 must be less than 0.5.\n'); return; end if f0 ~= 0 & f0 < 0.25/osr warning('(%s) Creating a lowpass ntf.', mfilename); f0 = 0; end if f0 ~= 0 & rem(order,2) ~= 0 fprintf(1,'Error. order must be even for a bandpass modulator.\n'); return; end if length(opt)>1 & length(opt)~=order fprintf(1,'The opt vector must be of length %d(=ord er).\n', order); return; end % Determine the zeros. if f0~=0 % Bandpass design-- halve the order tempor arily. order = order/2;

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dw = pi/(2*osr); else dw = pi/osr; end if length(opt)==1 if opt==0 z = zeros(order,1); else z = dw*ds_optzeros(order,1+rem(opt-1,2)); if isempty(z) return; end end if f0~=0 % Bandpass design-- shift and replicate th e zeros. order = order*2; z = z + 2*pi*f0; ztmp = [ z'; -z' ]; z = ztmp(:); end z = exp(j*z); else z = opt(:); end ntf = zpk(z,zeros(1,order),1,1); Hinf_itn_limit = 100; opt_iteration = 5; % Max number of zero-optimizing/Hinf iterations while opt_iteration > 0 % Iteratively determine the poles by finding the va lue of the x-parameter % which results in the desired H_inf. ftol = 1e-10; if f0>0.25 z_inf=1; else z_inf=-1; end if f0 == 0 % Lowpass design HinfLimit = 2^order; % !!! The limit is actually lower for opt=1 and low osr if H_inf >= HinfLimit fprintf(2,'%s warning: Unable to achieve specified Hinf.\n', mfilename); fprintf(2,'Setting all NTF poles to zero.\n'); ntf.p = zeros(order,1); else x=0.3^(order-1); % starting guess

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converged = 0; for itn=1:Hinf_itn_limit me2 = -0.5*(x^(2./order)); w = (2*[1:order]'-1)*pi/order; mb2 = 1+me2*exp(j*w); p = mb2 - sqrt(mb2.^2-1); out = find(abs(p)>1); p(out) = 1./p(out); % reflect poles to be inside the unit circle. p = cplxpair(p); ntf.z = z; ntf.p = p; f = real(evalTF(ntf,z_inf))-H_inf; % [ x f ] if itn==1 delta_x = -f/100; else delta_x = -f*delta_x/(f-fprev); end xplus = x+delta_x; if xplus>0 x = xplus; else x = x*0.1; &nbsp; end fprev = f; if abs(f)<ftol | abs(delta_x)<1e-10 converged = 1; break; end if x>1e6 fprintf(2, '%s warning: Unable to achieve specified Hinf.\n', mfilename); fprintf(2, 'Setting all NTF poles to zero.\n'); ntf.z = z; ntf.p = zeros(order,1); break; end if itn == Hinf_itn_limit fprintf(2,'%s warning: Danger! Iteration limit exce eded.\n', mfilename); end end end else % Bandpass design. x = 0.3^(order/2-1); % starting guess (not very good for f0~0) c2pif0 = cos(2*pi*f0); for itn=1:Hinf_itn_limit

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e2 = 0.5*x^(2./order); w = (2*[1:order]'-1)*pi/order; mb2 = c2pif0 + e2*exp(j*w); p = mb2 - sqrt(mb2.^2-1); % reflect poles to be inside the unit circle. out = find(abs(p)>1); p(out) = 1./p(out); p = cplxpair(p); ntf.z = z; ntf.p = p; f = real(evalTF(ntf,z_inf))-H_inf; % [x f] if itn==1 delta_x = -f/100; else delta_x = -f*delta_x/(f-fprev); end xplus = x+delta_x; if xplus > 0 x = xplus; else x = x*0.1; end fprev = f; if abs(f)<ftol | abs(delta_x)<1e-10 break; end if x>1e6 fprintf(2,'%s warning: Unable to achieve specified Hinf.\n', mfilename); fprintf(2,'Setting all NTF poles to zero.\n'); p = zeros(order,1); ntf.p = p; break; end if itn == Hinf_itn_limit fprintf(2,'%s warning: Danger! Hinf iteration limit exceeded.\n', mfilename); end end end %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% if opt < 3 % Do not optimize the zeros opt_iteration = 0; else zp = z(angle(z)>0); x0 = (angle(zp)-2*pi*f0) * osr / pi; if opt==4 & f0~=0 % Do not optimize the zeros at f0 x0(find( abs(x0)<1e-10 )) = []; end

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if f0 == 0 ub = ones(size(x0)); lb = zeros(size(x0)); else ub = 0.5*ones(size(x0)); lb = -ub; end options = optimset( 'TolX',0.001, 'TolFun',0.01, 'MaxIter',100 ); options = optimset(options, 'LargeScale','off'); options = optimset(options, 'Display','off'); %options = optimset(options,'Display','iter'); x = fmincon(@(x)ds_synNTFobj1(x,p,osr,f0),x0,[],[], [],[], lb,ub,[],options); z = exp(2i*pi*(f0+0.5/osr*x)); if f0>0 z = padt(z,length(p)/2,exp(2i*pi*f0)); end z = [z conj(z)]; z = z(:); if f0==0 z = padt(z,length(p),1); end ntf.z = z; ntf.p = p; if abs( real(evalTF(ntf,z_inf)) - H_inf ) < ftol opt_iteration = 0; else opt_iteration = opt_iteration - 1; end end end

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Appendix C Loop Filter Simulink Models

Figure 56 - First Order Loop Filter Simulink Model

Figure 57 - Second Order Loop Filter Simulink Model

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Figure 58 - Third Order Loop Filter Simulink Model

Figure 59 - Forth Order Loop Filter Simulink Model

Figure 60 - Fifth Order Loop Filter Simulink Model

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Appendix D DSP Simulation MATLAB Code

function [ input, output ] = DSPsim(f,fs,dur,ord,am p) global order thrsh gain a_H b_H a_G b_G u_G u_H h s h = fdesign.lowpass('N,Fc', 2, 30000, 2000000); lpf= design(h, 'butter'); order = ord; thrsh = 0.5; gain = 2; %Fifth order if(order==5) a_G = [1, -5, 10, -10, 5, -1]; b_G = [1, -4.192314285394014, 7.085800759102200 , -6.029580196237540, 2.581193648473645, -0.444444444020463]; a_H = [1, -4.192314285394014, 7.085800759102200 , -6.029580196237540, 2.581193648473645, -0.444444444020463]; b_H = [0.807685714605986, -2.914199240897800, 3 .970419803762460, -2.418806351526355, 0.555555555979537, 0]; %Fourth order elseif(order==4) a_G = [1, -4, 6, -4, 1]; b_G = [1, -3.194473669098806, 3.891959928172827 , -2.135788772313910, 0.444444297096043]; a_H = [1, -3.194473669098806, 3.891959928172827 , -2.135788772313910, 0.444444297096043]; b_H = [0.805526330901194, -2.108040071827174, 1 .864211227686091, -0.555555702903957, 0]; %Third order elseif(order==3) a_H = [1 -2.199 1.687 -0.4439]; b_H = [0.7985 -1.31 0.5561 0]; a_G = [1 -2.998 2.998 -1]; b_G = [1 -2.199 1.687 -0.4439]; %Second order elseif(order==2) a_H = [1 -1.225 0.4413]; b_H = [0.7774 -0.5587 0];

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a_G = [1 -1.999 1]; b_G = [1 -1.225 0.4413]; end %Create input signal t=linspace(0,dur,dur*fs); inBuf=amp*sin(2*pi*f*t); %Initialize variables u_G= zeros(1,order+1,'double'); u_H= zeros(1,order+1,'double'); h = 0; s = 0; sz=length(inBuf); outBuf=zeros(1,sz,'double'); %Modulate for i=1:1:length(inBuf) outBuf(i)=modulate(inBuf(i)); end %Filter lowpassed=filter(lpf,outBuf); %plot subplot(4,1,1), plot(t,inBuf); subplot(4,1,2), plot(t,outBuf); subplot(4,1,3), plot(t,lowpassed); subplot(4,1,4), semilogx(linspace(0,fs,length(t)),20*log10(abs(fft( lowpassed)))); %Generate signals for calcSNR input.signals.values=inBuf; input.time=t; output.signals.values=lowpassed; output.time=t; end function [y] = modulate(in) global order a_H b_H a_G b_G h s s=in-h; g=iir(s,a_G,b_G,0,order); y=quantize(g); h=iir(y,a_H,b_H,1,order); end function [y] = iir(in,a,b,buffer,order) y=0; global u_G u_H if(buffer==0) %u_G

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%circulate buffer for i=order+1:-1:2 u_G(i)=u_G(i-1); end u_G(1)=in*a(1); for i=2:1:order+1 u_G(1)= u_G(1) - (a(i)*u_G(i)); end for i=1:1:order+1 y=y+(b(i)*u_G(i)); end else %circulate buffer for i=order+1:-1:2 u_H(i)=u_H(i-1); end u_H(1)=in*a(1); for i=2:1:order+1 u_H(1)= u_H(1) - (a(i)*u_H(i)); end for i=1:1:order+1 y=y+(b(i)*u_H(i)); end end end function [y] = quantize(in) global thrsh gain if(in<-thrsh) y=-gain; elseif(in>thrsh) y=gain; else y=0.0; end end

Appendix E Coefficient Optimization MATLAB

Code

function [snr] = estimateCoeffs(coeffs) fs=2000000; %sampling frequency of 2Mhz dur=0.01; %duration %Parse the coeffs matrix into individual arrays of coefficients

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a_H = coeffs(1,:); b_H = coeffs(2,:); a_G = coeffs(3,:); b_G = coeffs(4,:);

%Determine the order of the given coefficients order=length(a_H)-1;

%Create a low pass filter for the output of the modulator h = fdesign.lowpass('N,Fc', 2, 30000, 2000000); Hd = design(h, 'butter');

%Generate a set of 20 frequencies between 1kHz and 20kHz f=linspace(1000,20000,20);

%For each frequency, do DSP simulation, and calculate the SNR of the output for i=1:1:length(f) [input, output] = DSPsimCoeffs(f(i),fs,dur,Hd, order, a_H,b_H,a_G,b_G); snrs(i)=newestCalcSNR(output,f(i),fs,0); end

%Take the average of all snr's and return it negate d (needed for fminsearch) snr=-(sum(snrs)/length(f));

function [result] = justdoit(mult)

tic; for i=1:1:5 order=i; %Fifth order if(order==5) a_G = [1, -5, 10, -10, 5, -1]; b_G = [1, -4.192314285394014, 7.08580075910 2200, -6.029580196237540, 2.581193648473645, -0.444444444020463]; a_H = [1, -4.192314285394014, 7.08580075910 2200, -6.029580196237540, 2.581193648473645, -0.444444444020463]; b_H = [0.807685714605986, -2.91419924089780 0, 3.970419803762460, -2.418806351526355, 0.555555555979537, 0]; %Fourth order elseif(order==4) a_G = [1, -4, 6, -4, 1]; b_G = [1, -3.194473669098806, 3.89195992817 2827, -2.135788772313910, 0.444444297096043]; a_H = [1, -3.194473669098806, 3.89195992817 2827, -2.135788772313910, 0.444444297096043]; b_H = [0.805526330901194, -2.10804007182717 4, 1.864211227686091, -0.555555702903957, 0]; %Thir order

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elseif(order==3) a_H = [1 -2.200261139334621 1.6886579756586 86 -0.444414218339989]; b_H = [0.799738860665379 -1.311342024341314 0.555585781660011 0]; a_G = [1 -3 3 -1]; b_G = [1 -2.200261139334621 1.6886579756586 86 -0.444414218339989]; %Second order elseif(order==2) a_H = [1 -1.225148226554415 0.441518440112 254]; b_H = [0.774851773445585 -0.55848155988774 6 0]; a_G = [1 -2 1]; b_G = [1 -1.225148226554415 0.441518440112 254]; %First order elseif (order==1) a_H = [1.000000000000000 -0.33333333335874 4]; b_H = [0.666666666641256 , 0]; a_G=[1 -1]; b_G=[1.000000000000000 -0.333333333358744] ; end %Do the magic result(i).order='Order'+i; result(i).oldcoeffs=[a_H;b_H;a_G;b_G]; result(i).oldsnr = -estimateCoeffs(result(i).ol dcoeffs); iter=mult*(4+i*4); options = optimset('MaxIter',iter,'MaxFunEvals' ,iter); [result(i).coefs,snr] = fminsearch(@estimateCoeffs,[a_H;b_H;a_G;b_G],option s); result(i).snr=-snr; end toc for i=1:1:5 x(i)=i; y1(i)=result(i).oldsnr; y2(i)=result(i).snr; end plot(x,y1,'--rs','LineWidth',2,... 'MarkerEdgeColor','k',... 'MarkerFaceColor','g',... 'MarkerSize',10); hold on; plot(x,y2,'--rs','LineWidth',2,... 'MarkerEdgeColor','b',... 'MarkerFaceColor','r',... 'MarkerSize',10); hold off;

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Appendix F Automated Testing MATLAB Code

getStats.m

function [snr,freq,thd,time] = getStats(fs) for j=1:1:3 for i=1:1:5 for n=1:1:18 [signal]=readFile(j,i,n); thd(j,i,n)=THD(signal,fs); tic; [snr(j,i,n),freq(j,i,n)] = optSNR(s ignal,fs); time(j,i,n)=toc; out=['SNR: ',num2str(snr(j,i,n))] out=['FREQ: ',num2str(freq(j,i,n))] out=['THD: ',num2str(thd(j,i,n))] out=['TIME: ',num2str(time(j,i,n))] end end end end function [signal] = readFile(j,i,n) switch j case 1 modFreq = '1MHz'; case 2 modFreq = '15MHz'; case 3 modFreq = '2MHz'; end name = ['.\cut_files\', modFreq, num2str((i-1)* 19+n+1),'.wav']; %name=['Ord ',num2str(i),' - ',num2str(n),'.wav '] signal = wavread(name); end function [thd1] = THD(output,fs) %freqs = fft(output.signals.values); y=output; L=length(y); NFFT = 2^nextpow2(L); % Next power of 2 from le ngth of y Y = fft(y,NFFT)/L; f = fs/2*linspace(0,1,NFFT/2); % Plot single-sided amplitude spectrum. plot(f,2*abs(Y(1:NFFT/2))) title('Single-Sided Amplitude Spectrum of y(t)' )

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xlabel('Frequency (Hz)') ylabel('|Y(f)|') %Number of harmonics harms = 5; Y=abs(Y); %Find index of fundamental frequency [temp,index]=max(Y); %Get magnitude of fundamental freq. fFund = Y(index); %Get magnitude of first 5 harmonics h1 = Y(2*index)^2; h2 = Y(3*index)^2; h3 = Y(4*index)^2; h4 = Y(5*index)^2; h5 = Y(6*index)^2; %Calculate their sum hSum = sqrt(h1+h2+h3+h4+h5); %Calculate THD thd1=(hSum/fFund)*100; end

optSNR.m

function [snr,f]= optSNR(output,fs) global x; global fs1; fs1=fs; x= output; f_opt = optF(output); [f,snr] = fminsearch(@calcSNR,f_opt); snr = -snr; end function f_opt = optF(output) global fs1; fDomain = abs(fft(output)); [c,i] = max(fDomain); f_opt = fs1*i/(length(fDomain)-1); end function [snr] = calcSNR(f) global x; global fs1; fs = fs1; r = 1; % A multiplyier that controls how much space is between truncations for b = 1:1:8*fs/f y = x((r*b):length(x)); n = [0:length(y)-1]'; A_hat=abs(y'*exp(-1j*2*pi*f*n/fs))/length(y)*2; theta_hat=atan(-(y'*sin(2*pi*f*n/fs)) / (y'*cos (2*pi*f*n/fs))); x_hat=A_hat*cos(2*pi*f*n/fs+theta_hat);

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% Time Domain Method w=y-x_hat; biggest(b) = max(w); end [smallest, i] = min(biggest); i = i(1); y = x((r*i):length(x)); n = (0:length(y)-1)'; A_hat=abs(y'*exp(-1j*2*pi*f*n/fs))/length(y)*2; theta_hat=atan(-(y'*sin(2*pi*f*n/fs)) / (y'*cos (2*pi*f*n/fs))); x_hat=A_hat*cos(2*pi*f*n/fs+theta_hat); % Time Domain Method w=y-x_hat; w1 = fft(w); %figure(1); plot(w); N1 = length(w1); wtrunc = w1(1:floor((length(w1)/fs)*20000)); sigPWR1 = x_hat'*x_hat; noisePWR1 = (wtrunc'*wtrunc)/(N1+1); snr = -10*log10(sigPWR1/noisePWR1); end

getZeroSNR.m

function [snr,freq,thd,time] = getZeroSNR(fs) for k=1:1:3 for j=1:1:3 for i=1:1:5 for n=1:1:19 [signal,noise]=readFile(k,j,i,n); [snr(k,j,i,n),freq(k,j,i,n)] = getS NR(signal,fs); time(k,j,i,n)=toc; out=[ 'SNR: ' ,num2str(snr(k,j,i,n))] out=[ 'FREQ: ' ,num2str(freq(k,j,i,n))] out=[ 'THD: ' ,num2str(thd(k,j,i,n))] out=[ 'TIME: ' ,num2str(time(k,j,i,n))] end end end end end function [signal,noise] = readFile(k,j,i,n) switch j case 1 modFreq = '1MHz' ; case 2 modFreq = '15MHz' ; case 3 modFreq = '2MHz' ;

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end signalName = [ '.\cut_files\' , modFreq, num2str((i-1)*19+n), '.wav' ]; noiseName = [ '.\zero input\' ,num2str(k), '.wav' ]; %name=['Ord ',num2str(i),' - ',num2str(n),'.wav'] signal = wavread(signalName); noise = wavread(noiseName); end function [snr,freq]= getSNR(signal,noise,fs) sig = abs(fft(signal)); freq = max(sig); sigPwr = max(sig)^2; noisePwr = sum(noise.^2); snr = 10*log10(sigPwr/noisePwr); end

format.m

function [page] = format(var) page = zeros(5*3,18); row=1; for i=1:1:3 for j=1:1:5 temp = var(i,j,:); page(row,:) = temp(:)'; row=row+1; end end

plotForUs.m

function plotForUs(var,xPlot,yPlot) [x, xLabel]=getData(var,xPlot); [y, yLabel]=getData(var,yPlot); subplot(3,1,1), myPlot(x,y,1,5); title('1Mhz Modulation','fontweight','b'); xlabel(xLabel); ylabel(yLabel); subplot(3,1,2), myPlot(x,y,6,10); title('1.5Mhz Modulation','fontweight','b'); xlabel(xLabel); ylabel(yLabel); subplot(3,1,3), myPlot(x,y,11,15); title('2Mhz Modulation','fontweight','b'); xlabel(xLabel); ylabel(yLabel); end

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function myPlot(x,y,first,last) for i=first:1:last plot(x(i,:),y(i,:),'-','Color',getColor(i), 'LineWidth',0.5); hold on; end hold off; legend('1st Order','2nd Order','3rd Order','4th Order','5th Order'); legend('show'); grid on; end function [color]= getColor(i) switch mod(i,5) case 1 color = [0.051 0.733 0.18]; case 2 color = [206/255 36/255 8/255]; case 3 color = [32/255 123/255 224/255]; case 4 color = [153/255 82/255 232/255]; case 0 color = [213/255 132/255 13/255]; end end function [out,label] = getData(var,plotVar) switch plotVar case 'snr' label='SNR (dB)'; out = var.snr; case 'freq' label = 'Frequency (Hz)'; out = var.freq; case 'thd' label = 'THD (%)'; out = var.thd; case 'time' label = 'Calc Time (s)'; out = var.time; end end

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Appendix G DSP Code

Main.c

/*********************************************** * * Class D Amplifier 2009 * ************************************************/ #include "tt.h" #include <stdio.h> #include <math.h> #include <SRU.h> #include <time.h> #include <sysreg.h> #define orders 5 //Defines the modulator order #define freqs 20 //used for testing 20 different f requencies #define amp 1.5 //quantizer gain #define N 296 //timer max count (Used for Timer IS R) #define G 0 //filter types #define H 1 #define min -2 // Min Output of quantizer #define max 2// Max output of quantizer #define thr_Lo -0.5// Threshold For Quantizer #define thr_Hi 0.5// Threshold for Quantizer #define time_diff 1.041666666666666666e-5 //Time increment for 96kHz #define time_diff2 0.0000005 //Time increment for 2 Mhz #define PI 3.14159265358979323846 /*Function declarations*/ double iir(double iirin, const double *a, const dou ble *b, double *u); double quantize(double inp, double low, double high ); void modulate(void); void clk(void); void initVars(void); void updateCoeffs(void); void doTriState(int status); /* Global variables */ int in,in2; bool flag,flag1,flag2; double x,x1,x2,x3,x4,s,h,g,y, adc, spdif; double a_H[6],b_H[6],a_G[6],b_G[6]; double freq[20];

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int order,orderPrev; double timeADC, timeFast; double u_H[6]; //H filter U buffer double u_G[6]; //G filter U buffer double adc_buf[2048],spdif_buf[1024]; int count; int freqNum; /*timer variables*/ volatile clock_t clock_start; volatile clock_t clock_stop; double secs,outf; int triState; void main(void) { //Initialize PLL to run at CCLK= 331.776 MHz & SDCLK= 165.888 MHz InitPLL_SDRAM(); // Need to initialize DAI because the sport sig nals need to be routed InitSPDIF(); // This function will configure the codec on th e kit Init1835viaSPI(); // Finally setup the sport to receive / transmi t the data InitSPORT(); order=orders; orderPrev=orders; //used for testing updateCoeffs(); //select coefficients based on order freqNum=0; //used for testing initVars(); //initialize all variables /*Create 20 different frequencies, starting a 2 0Hz, and incrementing by 1k 20 times. Used for testing only*/ freq[0]=20; int i; for(i=1;i<20;i++) { freq[i]=freq[i-1]+1000; } /*Setup input ISR (choose either adc or spdif)* / //interruptf(SIG_SP0,adcISR); interruptf(SIG_SP0,spdifISR); interruptf(SIG_TMZ0, timerISR); //enable high p riority timer interrupt timer_set(N, N); //set tperiod and tcount of th e timer timer_on(); //start timer while(1){ //Loop forever, interrupts will occur while this loop runs } }

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void initVars() { //initialize variables h=0.0; adc=0.0; s=0.0; g=0.0; y=0.0; count=0; flag2=0; flag1=0; flag=0; timeADC=0; outf=0; triState=2; int i; for (i=0;i< 6;i++) { u_H[i]=0.0; u_G[i]=0.0; } } void updateCoeffs(void) /*This function is used to choose the coefficients to use based on the order*/ { if(order==1) { double a_Hs[6]= { 1.050000000000000, -0.333333 333358744,

0,0,0,0}; double b_Hs[6]= { 0.666666666641256,

0,0,0,0,0}; double a_Gs[6]= {1.000000000000000, -1.000000 000000000,

0,0,0,0}; double b_Gs[6]= {1.000000000000000, -0.333333 333358744,

0,0,0,0}; int i=0; for(i;i<6;i++) { a_H[i]=a_Hs[i]; b_H[i]=b_Hs[i]; a_G[i]=a_Gs[i]; b_G[i]=b_Gs[i]; } } else if(order==2) { double a_Hs[6] = {1.000000000000000, -1.2251482 26554415,

0.441518440112254,0,0,0}; double b_Hs[6] = {0.774851773445585, - 0.558481559887746,

0,0,0,0}; double a_Gs[6] = {1.000000000000000, -2.0000000 00000000,

1.000000000000000,0,0,0}; double b_Gs[6] = { 1.050000000000000, -1.225148 226554415,

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0.441518440112254,0,0,0}; int i=0; for(i;i<6;i++) { a_H[i]=a_Hs[i]; b_H[i]=b_Hs[i]; a_G[i]=a_Gs[i]; b_G[i]=b_Gs[i]; } } else if(order==3) { double a_Hs[6] = {1,-2.2004, 1.6887514,-0.4444 1466}; double b_Hs[6] = {0.79974,-1.31157, 0.5556, 0 }; double a_Gs[6] = {1,-3,3,-1}; double b_Gs[6] = {1,-2.2004,1.6887514,-0.4444146 6}; int i=0; for(i;i<6;i++) { a_H[i]=a_Hs[i]; b_H[i]=b_Hs[i]; a_G[i]=a_Gs[i]; b_G[i]=b_Gs[i]; } } else if(order==4) { double a_Hs[6]={1.000000000000000, -3.194473669 098806,

3.891959928172827, -2.135788772313910, 0.444444297096043,0};

double b_Hs[6]={0.805526330901194, -2.108040071 827174, 1.864211227686091, - 0.555555702903957, 0,0};

double a_Gs[6]={1.050000000000000, -4.000000000 000000, 6.000000000000000, -4.000000000000000, 1.000000000000000,0};

double b_Gs[6]={1.000000000000000, -3.194473669 098806, 3.891959928172827, -2.135788772313910, 0.444444297096043,0};

int i=0; for(i;i<6;i++) { a_H[i]=a_Hs[i]; b_H[i]=b_Hs[i]; a_G[i]=a_Gs[i]; b_G[i]=b_Gs[i]; } } else if(order==5) { double a_Hs[6]= {1.000000000000000, -4.19231428 5394014,

7.085800759102200, -6.029580196237540, 2.581193648473645, -0.444444444020463};

double b_Hs[6]= {0.807685714605986, -2.91419924 0897800, 3.970419803762460, -2.418806351526355, 0.555555555979537, 0};

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115

double a_Gs[6]= {1.000000000000000, -5.00000000 0000000, 10.000000000000000, -10.000000000000000, 5.000000000000000, -1.000000000000000};

double b_Gs[6]= {1.000000000000000, -4.19231428 5394014, 7.085800759102200, -6.029580196237540, 2.581193648473645, -0.444444444020463};

int i=0; for(i;i<6;i++) { a_H[i]=a_Hs[i]; b_H[i]=b_Hs[i]; a_G[i]=a_Gs[i]; b_G[i]=b_Gs[i]; } } } /*********************************************** * * ISR for ADC Input * ************************************************/ void adcISR(int sig) { while((*pSPCTL0>>30)!=3) {} in2 = *pRXSP0A; in2 = *pRXSP0A; /*Used for generating test signals*/ timeADC+=time_diff; freqNum = timeADC/0.5; adc = cos(2*PI*freq[freqNum]*timeADC); /*Used for reading input in real time*/ //x1 = (double)in2; //adc = x1/ 2147483648.0; flag2=1; } void spdifISR(int sig) { sysreg_bit_clr(sysreg_LIRPTL,SP4IMSK); /*Check if input is ready to avoid core hand*/ while((*pSPCTL0>>30)!=3) {} /*Acquire input sample from SP0A register (must be done twice for

stability)*/ in = *pRXSP0A; in = *pRXSP0A; x2 = (double)in; spdif = x2/ 2147483648.0; /*Use this if input is needed to be buffered for a nalysis*/ /*

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116

spdif_buf[count]=spdif; if (count==1024) count=count;*/ sysreg_write(sysreg_LIRPTL,SP4IMSK); } void modulate(void) { //clock_start=clock(); //used for timing //s = adc - h; //used with adc input /*Subtract previous h from input*/ s = spdif - h; /*G filter*/ g = iir(s,a_G,b_G,u_G); /*Quantize output of G filter*/ y = quantize(g,thr_Lo, thr_Hi); /*H filter*/ h = iir(y,a_H,b_H,u_H); /*Use this for timing*/ //clock_stop=clock(); //secs = ((double) (clock_stop - clock_start)) / C LOCKS_PER_SEC; //printf("Time taken is %e seconds\n",secs); flag1=0; } /*********************************************** * * Timer ISR * ************************************************/ void timerISR(int sig) { doTriState(triState); //Update digital outputs /*This is used for testing*/ /* if(freqNum==19) { freqNum=0; timeADC=0; //order++; } if(order>5) order=3; //break here if(order!=orderPrev) //if the order has changed { initVars(); updateCoeffs(); }

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117

orderPrev=order; */ modulate(); } /*********************************************** * * Quantizer * ************************************************/ double quantize(double inp, double low, double high ) { int out; if((inp<low)&&(!flag)) { out = min; triState=0; } else if((inp>high)&&(!flag)) { out = max; triState=1; } else { out = 0.0; triState=2; } return out; } void doTriState(int status) { if(triState==0) { SRU(LOW,DAI_PB16_I); //turn off LED 7 SRU(HIGH,DAI_PB15_I); //light LED 6 } else if(triState==1) { SRU(HIGH,DAI_PB16_I); //light LED 7 SRU(LOW,DAI_PB15_I); //turn off LED 6 } else if(triState==2) { SRU(LOW,DAI_PB16_I); //turn off LED 7 SRU(LOW,DAI_PB15_I); //turn off LED 6 } } /*********************************************** * * General IIR Filter *

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118

************************************************/ double iir(double iirin, const double *a, const dou ble *b, double *u) { int i; double out=0; /*Circulate buffer*/ for(i=order; i>0; i--) u[i]=u[i-1]; u[0] = iirin*a[0]; /*Tap with a coefficients*/ for(i=1;i<=order;i++) u[0]-=a[i]*u[i]; /*Tap with b coefficients*/ for(i=0;i<=order;i++) out+= b[i]*u[i]; return out; } init1835viaSPI.c

/////////////////////////////////////////////////////////////////////////////////////// //NAME: init1835viaSPI.c (Block-based Talkthrough) //DATE: 7/29/05 //PURPOSE: Talkthrough framework for sending and receiving samples to the AD1835. // //USAGE: This file contains the subroutines for accessing the AD1835 control // registers via SPI. // //////////////////////////////////////////////////////////////////////////////////////// #include "tt.h" #include "ad1835.h" /* Setup the SPI pramaters here in a buffer first */ unsigned int Config1835Param [] = { WR | DACCTRL1 | DACI2S | DAC24BIT | DACFS96, WR | DACCTRL2 ,//| DACMUTE_R4 | DACMUTE_L4, WR | DACVOL_L1 | DACVOL_MAX, WR | DACVOL_R1 | DACVOL_MAX, WR | DACVOL_L2 | DACVOL_MAX, WR | DACVOL_R2 | DACVOL_MAX, WR | DACVOL_L3 | DACVOL_MAX, WR | DACVOL_R3 | DACVOL_MAX, WR | DACVOL_L4 | DACVOL_MAX, WR | DACVOL_R4 | DACVOL_MAX, WR | ADCCTRL1 | ADCFS96, WR | ADCCTRL2 | ADCI2S | ADC24BIT, WR | ADCCTRL3 | IMCLKx2 } ; volatile int spiFlag ;

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119

//Set up the SPI port to access the AD1835 void SetupSPI1835 () { /* First configure the SPI Control registers */ /* First clear a few registers */ *pSPICTL = (TXFLSH | RXFLSH) ; *pSPIFLG = 0; *pSPICTL = 0; /* Setup the baud rate to 500 KHz */ *pSPIBAUD = 100; /* Setup the SPI Flag register to FLAG3 : 0xF708*/ *pSPIFLG = 0xF708; /* Now setup the SPI Control register : 0x5281*/ *pSPICTL = (SPIEN | SPIMS | MSBF | WL16 | TIMOD1) ; } //Disable the SPI Port void DisableSPI1835 () { *pSPICTL = (TXFLSH | RXFLSH); } //Send a word to the AD1835 via SPI void Configure1835Register (int val) { *pTXSPI = val ; Delay(100); //Wait for the SPI to indicate that it has finished. while (1) { if (*pSPISTAT & SPIF) break ; } Delay (100) ; } //Receive a register setting from the AD1835 unsigned int Get1835Register (int val) { *pTXSPI = val ; Delay(100); //Wait for the SPI port to indicate that it has finished while (1) { if (SPIF & *pSPISTAT) break ; } Delay (100) ; return *pRXSPI ; // return i ; } //Set up all AD1835 registers via SPI void Init1835viaSPI() { int configSize = sizeof (Config1835Param) / sizeof (int) ; int i ;

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120

SetupSPI1835 () ; for (i = 0; i < configSize; ++i) { Configure1835Register (Config1835Param[i]) ; } DisableSPI1835 () ; } //Delay loop void Delay (int i) { for (;i>0;--i) asm ("nop;") ; }

initPLL_SDRAM.c

/*************************************************************************************** ** ** File: initPLL.c ** Date: 7-29-05 ** Author: SH ** Use: Initialize the DSP PLL for the required CCLK and HCLK rates. ** Note: CLKIN will be 24.576 MHz from an external oscillator. The PLL is ** programmed ** to generate a core clock (CCLK) of 331.776 MHz - PLL multiplier = 27 and ** divider = 2. ** *************************************************************************************/ #include <def21369.h> #include <cdef21369.h> void InitPLL_SDRAM(){ /************************************************************************************/ int i, pmctlsetting; //Change this value to optimize the performance for quazi-sequential accesses //(step > //1) #define SDMODIFY 1 pmctlsetting= *pPMCTL; pmctlsetting &= ~(0xFF); //Clear // CLKIN= 24.576 MHz, Multiplier= 27, Divisor= 2, CCLK_SDCLK_RATIO 2. // Core clock = (24.576 MHz * 27) /2 = 331.776 MHz pmctlsetting= SDCKR2|PLLM32|PLLD2|DIVEN; *pPMCTL= pmctlsetting; pmctlsetting|= PLLBP; *pPMCTL= pmctlsetting; //Wait for around 4096 cycles for the pll to lock. for (i=0; i<4096; i++) asm("nop;"); *pPMCTL ^= PLLBP; //Clear Bypass Mode

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121

*pPMCTL |= (CLKOUTEN); //and start clkout // Programming SDRAM control registers and enabling SDRAM read optimization // CCLK_SDCLK_RATIO= 2.5 // RDIV = ((f SDCLK X t REF )/NRA) - (tRAS + tRP ) // (166*(10^6)*64*(10^-3)/4096) - (7+3) = 2583 *pSDRRC= (0xA17)|(SDMODIFY<<17)|SDROPT; //=================================================================== // // Configure SDRAM Control Register (SDCTL) for PART MT48LC4M32B2 // // SDCL3 : SDRAM CAS Latency= 3 cycles // DSDCLK1: Disable SDRAM Clock 1 // SDPSS : Start SDRAM Power up Sequence // SDCAW8 : SDRAM Bank Column Address Width= 8 bits // SDRAW12: SDRAM Row Address Width= 12 bits // SDTRAS7: SDRAM tRAS Specification. Active Command delay = 7 cycles // SDTRP3 : SDRAM tRP Specification. Precharge delay = 3 cycles. // SDTWR2 : SDRAM tWR Specification. tWR = 2 cycles. // SDTRCD3: SDRAM tRCD Specification. tRCD = 3 cycles. // //-------------------------------------------------------------------- *pSDCTL= SDCL3|DSDCLK1|SDPSS|SDCAW8|SDRAW12|SDTRAS7|SDTRP3|SDTWR2|SDTRCD3; // Note that MS2 & MS3 pin multiplexed with flag2 & flag3. // MSEN bit must be enabled to access SDRAM, but LED7 cannot be driven with sdram *pSYSCTL |=MSEN; // Mapping Bank 2 to SDRAM // Make sure that jumper is set appropriately so that MS2 is connected to // chip select of 16-bit SDRAM device *pEPCTL |=B2SD; *pEPCTL &= ~(B0SD|B1SD|B3SD); //=================================================================== // // Configure AMI Control Register (AMICTL0) Bank 0 for the ISSI IS61LV5128 // // WS2 : Wait States = 2 cycles // HC1 : Bus Hold Cycle (at end of write access)= 1 cycle. // AMIEN: Enable AMI // BW8 : External Data Bus Width= 8 bits. // //-------------------------------------------------------------------- //SRAM Settings *pAMICTL0 = WS2|HC1|AMIEN|BW8; //=================================================================== // // Configure AMI Control Register (AMICTL) Bank 1 for the AMD AM29LV08 // // WS23 : Wait States= 23 cycles // AMIEN: Enable AMI // BW8 : External Data Bus Width= 8 bits. // //-------------------------------------------------------------------- //Flash Settings

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122

*pAMICTL1 = WS23|AMIEN|BW8; }

initSPORT.c

////////////////////////////////////////////////////////////////////////////////////// //NAME: initSPORT.c (Block-based Talkthrough) //DATE: 7/29/05 //PURPOSE: Talkthrough framework for sending and receiving samples to the AD1835. // //USAGE: This file uses SPORT0 to receive data from the ADC and transmits the // data to the DAC's via SPORT1A, SPORT1B, SPORT2A and SPORT2B. // DMA Chaining is enabled // ////////////////////////////////////////////////////////////////////////////////////// #include "tt.h" /* Here is the mapping between the SPORTS and the DACS ADC -> DSP : SPORT0A : I2S DSP -> DAC1 : SPORT1A : I2S DSP -> DAC2 : SPORT1B : I2S DSP -> DAC3 : SPORT2A : I2S DSP -> DAC4 : SPORT2B : I2S */ void InitSPORT() { //Clear the Mutlichannel control registers *pSPMCTL0 = 0; *pSPMCTL1 = 0; *pSPMCTL2 = 0; *pSPCTL0 = 0 ; *pSPCTL1 = 0 ; *pSPCTL2 = 0 ; //============================================================ // // Configure SPORT 0 for input from SPDIF // //------------------------------------------------------------ *pSPCTL0 = (OPMODE | SLEN32 | SPEN_A); //============================================================ // // Configure SPORT2B for output to DAC 4 // //------------------------------------------------------------ *pSPCTL2 = SPTRAN | OPMODE | SLEN32 | SPEN_B; } initSRU.c

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123

void InitSRU(){ // Disable the pull-up resistors on all 20 pins *pDAI_PIN_PULLUP = 0x000FFFFF; // Set up SPORT 0 to receive from the SPDIF receiver // Tie the pin buffer input LOW. SRU(LOW,DAI_PB18_I); // Tie the pin buffer enable input LOW SRU(LOW,PBEN18_I); // Connect the SPDIF Receiver SRU(DAI_PB18_O,DIR_I); // Clock in from SPDIF RX SRU(DIR_CLK_O,SPORT0_CLK_I); // Frame sync from SPDIF RX SRU(DIR_FS_O,SPORT0_FS_I); // Data in from SPDIF RX SRU(DIR_DAT_O,SPORT0_DA_I); // Clock on pin 7 SRU(DIR_CLK_O,DAI_PB07_I); // Frame sync on pin 8 SRU(DIR_FS_O,DAI_PB08_I); // Tie the pin buffer enable inputs HIGH to drive DAI pins 7 and 8 SRU(HIGH,PBEN07_I ); SRU(HIGH,PBEN08_I ); //----------------------------------------------------------------------------- // // Connect the DACs: The codec accepts a BCLK input from DAI pin 13 and // a LRCLK (a.k.a. frame sync) from DAI pin 14 and has four // serial data outputs to DAI pins 12, 11, 10 and 9 // // Connect DAC1 to SPORT1, using data output A // Connect DAC2 to SPORT1, using data output B // Connect DAC3 to SPORT2, using data output A // Connect DAC4 to SPORT2, using data output B // // Connect MCLK from SPDIF to DAC on DAI Pin 6 // // Connect the clock and frame sync inputs to SPORT1 and SPORT2 // should come from the SPDIF RX on DAI pins 7 and 8, respectively // // Connect the SPDIF RX BCLK and LRCLK out to the DAC on DAI // pins 13 and 14, respectively. // // All six DAC connections are always outputs from the SHARC // so tie the pin buffer enable inputs all high. // //------------------------------------------------------------------------ //------------------------------------------------------------------------ // Connect the pin buffers to the SPORT data lines

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124

SRU(SPORT2_DB_O,DAI_PB09_I); SRU(SPORT2_DA_O,DAI_PB10_I); SRU(SPORT1_DB_O,DAI_PB11_I); SRU(SPORT1_DA_O,DAI_PB12_I); SRU(LOW,SPORT2_DB_I); SRU(LOW,SPORT2_DA_I); SRU(LOW,SPORT1_DB_I); SRU(LOW,SPORT1_DA_I); //------------------------------------------------------------------------ // Connect the clock, frame sync, and MCLK from the SPDIF RX directly // to the output pins driving the DACs. SRU(DIR_CLK_O,DAI_PB13_I); SRU(DIR_FS_O,DAI_PB14_I); SRU(DIR_TDMCLK_O,DAI_PB06_I); //------------------------------------------------------------------------ // Connect the SPORT clocks and frame syncs to the clock and // frame sync from the SPDIF receiver SRU(DIR_CLK_O,SPORT1_CLK_I); SRU(DIR_CLK_O,SPORT2_CLK_I); SRU(DIR_FS_O,SPORT1_FS_I); SRU(DIR_FS_O,SPORT2_FS_I); //------------------------------------------------------------------------ // Tie the pin buffer enable inputs HIGH to make DAI pins 9-14 outputs. SRU(HIGH,PBEN06_I); SRU(HIGH,PBEN09_I); SRU(HIGH,PBEN10_I); SRU(HIGH,PBEN11_I); SRU(HIGH,PBEN12_I); SRU(HIGH,PBEN13_I); SRU(HIGH,PBEN14_I); //-------------------------------------------------------------------------- // Route SPI signals to AD1835. SRU(SPI_MOSI_O,DPI_PB01_I) //Connect MOSI to DPI PB1. SRU(DPI_PB02_O, SPI_MISO_I) //Connect DPI PB2 to MISO. SRU(SPI_CLK_O, DPI_PB03_I) //Connect SPI CLK to DPI PB3. SRU(SPI_FLG3_O, DPI_PB04_I) //Connect SPI FLAG3 to DPI PB4. //--------------------------------------------------------------------------- // Tie pin buffer enable from SPI peripherals to determine whether they are // inputs or outputs SRU(SPI_MOSI_PBEN_O, DPI_PBEN01_I); SRU(SPI_MISO_PBEN_O, DPI_PBEN02_I); SRU(SPI_CLK_PBEN_O, DPI_PBEN03_I); SRU(SPI_FLG3_PBEN_O, DPI_PBEN04_I); //----------------------------------------------------------------------------- *pDIRCTL=0x0; }

Page 136: DSP Modulated Class D Audio Amplifier

Appendix H Raw Zero Input SNR Data

Calculated Input Frequency [Hz]

Mod.

Order 1021.380637 2011.180398 3006.809189 4048.443791 5055.516755 6062.600132 7130.909535 8076.741181 9083.819801

Ze

ro I

np

ut

SN

R [

dB

]

Fs

= 1

MH

z

1 109.6811959 108.4170688 107.9050478 109.6557744 109.7792657 107.8489634 110.2122014 108.0521061 109.8451153

2 110.1311723 108.8479978 108.5200213 109.9312065 110.1845344 108.2209902 110.5888429 107.8665156 109.9766897

3 110.0725683 109.09755 108.1500011 110.2297207 110.0640865 108.6649729 110.614034 107.3225259 110.2584778

5 110.1270384 108.9845803 108.3453092 110.0920354 110.2207978 108.271864 110.6412766 107.9287235 110.0522568

5 108.551514 109.2113906 108.1786679 110.2630905 110.1036717 108.6982807 110.6507609 107.3690793 110.2927466

Fs

= 1

.5M

Hz 1 109.6436816 108.4088908 107.8920751 109.6551223 109.7658771 107.8675555 110.234399 108.0721001 109.8819995

2 110.1335569 108.8658444 108.5347095 109.924806 110.1642235 108.1969863 110.5382637 107.8015119 109.8926668

3 110.1015534 109.1141757 108.1505234 110.2368385 110.073334 108.6659973 110.6151487 107.3048251 110.2380462

4 110.1525886 109.0008258 108.3540794 110.1166196 110.241971 108.3097763 110.6759248 107.9679341 110.1101024

5 110.0917623 109.1097346 108.1545126 110.2495396 110.0792114 108.6872848 110.6459237 107.3657432 110.3193945

Fs

= 2

MH

z

1 109.5505667 108.3597229 107.8627106 109.6301365 109.7708343 107.8328247 110.2098826 108.0549283 109.8487934

2 110.130457 108.8384793 108.4916349 109.9082415 110.1636481 108.2027451 110.5677171 107.8521573 109.9576154

3 110.0709215 109.0821841 108.1275152 110.2214812 110.0630902 108.6608418 110.6139757 107.322337 110.2577079

4 110.1173099 108.9673497 108.3244406 110.0821602 110.2039593 108.2604342 110.6262016 107.9138531 110.0411029

5 110.0856022 109.1067824 108.1416205 110.231447 110.066583 108.6630319 110.6232027 107.3221021 110.2719308

Calculated Input Frequency [Hz]

Mod.

Order 10090.89701 11097.97306 12202.32574 13112.1247 14119.20544 15126.2802 16133.35859 17183.57377 18147.50825

Ze

ro I

np

ut

SN

R [

dB

]

Fs

= 1

MH

z

1 108.7852919 108.5684128 109.7349073 106.3802404 109.2428226 108.2057232 107.6784247 108.9313795 106.1825289

2 109.5111204 108.2771771 110.0069844 106.6198588 109.6550082 107.6196064 108.5106747 108.5547364 106.3094278

3 109.6000112 108.3745181 110.1199565 106.7478297 109.5832149 108.5209021 107.9609186 109.0598234 105.0398678

5 109.6096358 108.3899086 110.1478109 106.7947124 109.6426688 108.6077244 108.0658845 108.8427453 106.6243425

5 109.2298222 109.0081121 110.1743367 106.8198671 109.6735463 108.6182088 108.1043592 109.2311933 105.2713385

Fs

= 1

.5M

Hz 1 108.824477 108.6227728 109.7840039 106.4328469 109.3106368 108.2691141 107.7421987 109.0157113 106.2807087

2 109.4101881 108.1487483 109.8638793 106.4416991 109.4572272 107.3831463 108.2574686 108.2755911 106.0093903

3 109.5598867 108.3122045 110.0223072 106.6111622 109.3999943 108.2712339 107.6436203 108.6561111 104.5443889

4 109.6771095 108.4772578 110.2417559 106.8900626 109.7477061 108.692733 108.136279 108.8647152 106.5868251

5 109.2777558 109.0971447 110.3072078 107.0105673 109.9454137 108.9907119 108.5524801 109.762346 105.8067404

Fs

= 2

MH

z

1 108.7915357 108.5794316 109.7520634 106.3940431 109.2652827 108.2273733 107.7028774 108.9563514 106.2125422

2 109.4952218 108.2555675 109.9909702 106.605064 109.6353085 107.599669 106.4308875 108.5362034 106.2998792

3 109.6029164 108.371615 110.1180231 106.7449161 109.5883613 108.5186501 107.9680163 109.0585921 105.0380087

4 109.5964571 108.3783126 110.1316601 106.7707121 109.6263602 108.5839466 108.0550368 108.8148808 106.6100713

5 109.2083875 108.9746173 110.1505187 106.7919852 109.6456585 108.6111372 108.0936657 109.2303444 105.2617975

Page 137: DSP Modulated Class D Audio Amplifier

Appendix I Raw SNR Data

Calculated Input Frequency [Hz]

Mod. Order 1021.380637 2011.180398 3006.809189 4048.443791 5055.516755 6062.600132 8076.741181 9083.819801

SN

R [

dB

]

Fs

= 1

MH

z 1 45.11185579 45.1283063 44.57499079 44.74611476 44.48741657 43.33566513 42.03667766 39.82854259

2 48.52110592 49.84604009 49.50754583 48.94168831 47.59261534 46.67456707 44.81198711 41.38742344

3 48.67642241 50.48020152 50.05277782 48.84496852 47.68242669 46.68420422 44.39861298 41.42024456

4 49.93942597 49.3985185 48.10052365 46.84680781 45.29959701 44.08988301 42.19800099 39.55067913

5 39.67618539 20.38901237 37.36827115 36.01869444 35.02029214 34.02292038 32.72307649 32.03488525

Fs

= 1

.5M

Hz

1 42.01622108 40.02929696 39.14195948 39.26291878 37.28515753 36.73752466 35.90380529 34.55846934

2 46.6359909 45.31614001 43.91373786 42.31473878 40.93525846 39.48989248 37.40336365 35.8221983

3 49.54748641 46.82188055 45.13279777 42.9409379 41.5046868 39.96659701 37.80528285 36.22771631

4 45.19656196 44.33523051 43.08824367 41.7985055 40.49269606 39.29852906 37.32390467 35.72588541

5 42.41109909 41.59930343 41.10999433 40.10143825 39.03726705 37.89730429 36.21006825 35.01824908

Fs

= 2

MH

z 1 45.64372969 45.25181425 44.47678009 44.86772662 44.48387615 43.70013119 42.72760648 40.29678721

2 49.9884773 49.95813394 49.97302048 49.0411312 47.96898856 47.10328637 45.67171538 41.78203038

3 51.06688962 50.75162027 50.3210815 49.46055811 48.28433733 47.34136958 45.08828145 41.56424127

4 48.99619662 47.40696298 45.8454424 44.3732176 42.94515873 41.67705799 39.7758403 37.77490722

5 47.8661796 45.87257717 43.58606267 41.95979008 40.24123863 38.89526349 36.63734673 35.35195263

Calculated Input Frequency [Hz]

Mod. Order 10090.89701 11097.97306 13112.1247 14119.20544 15126.2802 16133.35859 18147.50825

SN

R [

dB

]

Fs

= 1

MH

z 1 40.66938466 40.11916614 37.72239743 37.29532094 36.66082765 34.22067851 32.84548213

2 42.2812354 42.02694197 38.55368715 38.08962389 37.30955833 34.91913909 33.17488866

3 42.22739531 41.60872717 38.52103184 37.68199409 37.11743204 34.77137347 33.02247022

4 39.95134157 39.3145724 36.46741191 36.06176293 35.63339136 33.46482288 31.63662214

5 35.30038062 34.61102677 32.67494166 32.14343882 31.56534833 31.16371781 29.8145695

Fs

= 1

.5M

Hz

1 35.10853828 34.0891824 32.81302988 32.08145258 31.72036182 30.72799625 29.73643597

2 36.49919737 35.51318765 33.4911325 32.69768548 31.88882609 31.22739526 30.03764716

3 36.76988843 35.88391395 33.79473596 32.89700974 32.18071099 31.31167996 29.93089446

4 36.60633641 35.82349632 33.80657574 33.18880957 32.32690477 31.67940938 30.00983463

5 35.88712666 35.20260688 33.40034301 32.87123674 32.19609142 31.82077719 30.53577605

Fs

= 2

MH

z 1 40.49892246 39.94296068 36.07862889 35.6631873 35.02538852 33.98347548 32.38952531

2 42.31302908 41.69394297 36.91929319 36.06245578 35.14009325 6.365561313 32.93939297

3 41.89209581 41.34109078 36.46673816 35.85621443 35.09703022 34.14171796 32.43354893

4 38.41364068 37.89530949 34.36401473 33.71370442 32.67250404 31.91225108 30.46235922

5 36.97095605 36.08317943 33.60201221 32.89738264 32.20051836 31.38515246 30.14309011

Page 138: DSP Modulated Class D Audio Amplifier

Appendix J Raw THD Data

Calculated Input Frequency [Hz]

Mod. Order 1021.380637 2011.180398 3006.809189 4048.443791 5055.516755 6062.600132 8076.741181 9083.819801

TH

D [

%]

Fs

= 1

MH

z 1 0.595839564 0.210036637 0.167681573 0.25510573 0.330239785 0.254963466 0.198988689 0.155521664

2 0.642315698 0.273972212 0.300281601 0.255184086 0.273160215 0.225934107 0.124619787 0.09142683

3 0.576708512 0.261997858 0.094688025 0.194661044 0.247684062 0.190985383 0.11587523 0.120361126

5 0.513236589 0.258545981 0.20707789 0.421656348 0.454820321 0.297869693 0.333565415 0.289440518

5 0.611272762 0.7819743 0.291146511 2.221477614 2.235278736 2.344135475 2.063564628 1.941519167

Fs

= 1

.5M

Hz

1 0.592220948 0.294553535 0.315758524 0.21520284 0.675011374 0.394869601 0.13174512 0.168508828

2 0.605277116 0.274229744 0.153296515 0.214278641 0.25211169 0.159991036 0.133065656 0.161652004

3 0.606690224 0.253659329 0.233309626 0.169469166 0.27335408 0.167850853 0.210610278 0.193810264

4 0.609920048 0.258200601 0.292791603 0.145158168 0.301547474 0.281387241 0.208925207 0.253158481

5 0.637072965 0.239267247 0.238479613 0.298650742 0.419411385 0.316351905 0.418865267 0.297551428

Fs

= 2

MH

z 1 0.551950237 0.278757244 0.249300928 0.257268057 0.443992342 0.080091682 0.250565478 0.121954907

2 0.646676239 0.271400445 0.283237132 0.231930366 0.340603269 0.246760626 0.068372137 0.153521964

3 0.588155243 0.221936155 0.295009838 0.288549141 0.103363168 0.166266046 0.091777081 0.169056606

4 0.660338914 0.265421734 0.237723196 0.664798118 0.517642692 0.610229266 0.664266902 0.598589308

5 0.728995341 0.215158924 0.201649579 0.795031303 0.930970083 1.121075334 1.210670802 1.090125027

Calculated Input Frequency [Hz]

Mod. Order 10090.89701 11097.97306 13112.1247 14119.20544 15126.2802 16133.35859 18147.50825

TH

D [

%]

Fs

= 1

MH

z 1 0.144252938 0.114893785 0.096079322 0.177254248 0.150393157 0.107295422 0.171374893

2 0.074107754 0.125552106 0.098478809 0.113615559 0.128547883 0.09128277 0.151202445

3 0.131682253 0.090100616 0.068203648 0.094873495 0.083374717 0.106448278 0.143572194

5 0.291480844 0.236487136 0.156439981 0.149975352 0.126378335 0.182020218 0.205196706

5 1.596007168 1.366384252 0.687822199 0.419073763 0.122119255 0.124232019 0.330840696

Fs

= 1

.5M

Hz

1 0.126368359 0.191749032 0.2155077 0.235604872 0.097832233 0.158438504 0.130392319

2 0.103702829 0.189606084 0.134036555 0.192298529 0.130371744 0.113015111 0.154499559

3 0.159089053 0.197214554 0.172642818 0.193562418 0.213520373 0.199969486 0.178334979

4 0.228513686 0.266649252 0.14680173 0.271029539 0.236868867 0.135253512 0.277786764

5 0.201450061 0.399279449 0.248959677 0.236346661 0.245537648 0.149689537 0.247485814

Fs

= 2

MH

z 1 0.0987823 0.160845388 0.128247895 0.128681362 0.042391419 0.112743292 0.131550531

2 0.135956516 0.092565682 0.1227844 0.170120242 0.083570373 0.270310642 0.202953122

3 0.11220544 0.073428429 0.091723296 0.096713236 0.121612743 0.122592347 0.170781935

4 0.568751276 0.429900091 0.199252527 0.212066967 0.199579787 0.11626091 0.166366855

5 0.972642312 0.800839034 0.574013041 0.345858419 0.155208077 0.16206899 0.319165873

Page 139: DSP Modulated Class D Audio Amplifier

128

Appendix K Raw Power and Efficiency Data Input Freq.

[Hz]

Mod. Freq.

[MHz]

Mod.

Order

Input

Amplitude

Vin

[Volts]

I in

[Amps]

Pin

[watts]

Rout

[Ohms]

Vout

(RMS) [V]

Pout

[watts] Efficiency

Efficiency

[%] Average Efficiency

20 2 1 2 39.3 2.16 84.888 8.07 25.85 82.8033 0.975441567 97.54416

100 2 1 2 39.7 2.19 86.943 8.07 26.08 84.2833 0.969408934 96.94089

800 2 1 2 39.7 2.2 87.34 8.07 26.15 84.7364 0.97018971 97.01897

1500 2 1 2 39.7 2.18 86.546 8.07 26.04 84.025 0.970870767 97.08708

4000 2 1 2 39.7 2.19 86.943 8.07 26.2 85.0607 0.978350399 97.83504

12000 2 1 2 39.7 1.97 78.209 8.07 24.85 76.5208 0.978413685 97.84137

20000 2 1 2 39.8 1.45 57.71 8.07 21 54.6468 0.946921507 94.69215 0.969942367

20 1.5 1 2 39.1 2.34 91.494 8.07 27.05 90.6695 0.990987986 99.0988

100 1.5 1 2 39.6 2.4 95.04 8.07 27.44 93.3028 0.981721386 98.17214

800 1.5 1 2 39.6 2.39 94.644 8.07 27.55 94.0524 0.993748726 99.37487

1500 1.5 1 2 39.6 2.4 95.04 8.07 26.93 89.8668 0.945567952 94.5568

4000 1.5 1 2 39.6 2.4 95.04 8.07 26.74 88.6032 0.932272435 93.22724

12000 1.5 1 2 39.6 2.22 87.912 8.07 26.26 85.4508 0.97200332 97.20033

20000 1.5 1 2 39.7 1.79 71.063 8.07 22.92 65.0962 0.916035183 91.60352 0.961762427

20 1 1 2 39.3 2.15 84.495 8.07 25.74 82.1001 0.971656007 97.1656

100 1 1 2 39.4 2.17 85.498 8.07 25.76 82.2277 0.961750071 96.17501

800 1 1 2 39.4 2.17 85.498 8.07 25.92 83.2523 0.97373438 97.37344

1500 1 1 2 39.4 2.17 85.498 8.07 25.81 82.5472 0.965487196 96.54872

4000 1 1 2 39.4 2.17 85.498 8.07 25.93 83.3166 0.974485863 97.44859

12000 1 1 2 39.5 1.94 76.63 8.07 24.44 74.0166 0.965895278 96.58953

20000 1 1 2 39.6 1.43 56.628 8.07 20.79 53.5594 0.945810695 94.58107 0.965545641

20 2 2 2 38.1 2.31 88.011 8.07 26.19 84.9958 0.965740638 96.57406

100 2 2 2 39.5 2.4 94.8 8.07 27.32 92.4885 0.975617357 97.56174

800 2 2 2 39.7 2.41 95.677 8.07 27.4 93.031 0.97234423 97.23442

1500 2 2 2 39.7 2.41 95.677 8.07 27.48 93.575 0.978030442 97.80304

4000 2 2 2 39.8 2.41 95.918 8.07 27.32 92.4885 0.964245766 96.42458

12000 2 2 2 39.8 2.12 84.376 8.07 25.65 81.527 0.966233901 96.62339

20000 2 2 2 39.8 1.49 59.302 8.07 21.17 55.5352 0.93648072 93.64807 0.965527579

20 1.5 2 2 38.1 2.31 88.011 8.07 26.26 85.4508 0.970909953 97.091

100 1.5 2 2 39.6 2.41 95.436 8.07 27.41 93.0989 0.975511308 97.55113

800 1.5 2 2 39.9 2.44 97.356 8.07 27.63 94.5994 0.971685033 97.1685

1500 1.5 2 2 39.9 2.44 97.356 8.07 27.65 94.7364 0.973092252 97.30923

4000 1.5 2 2 39.9 2.44 97.356 8.07 27.55 94.0524 0.966066338 96.60663

12000 1.5 2 2 40 2.1 84 8.07 25.35 79.631 0.947988582 94.79886

20000 1.5 2 2 39.9 1.47 58.653 8.07 20.93 54.2831 0.92549631 92.54963 0.961535682

20 1 2 2 38 2.29 87.02 8.07 26.29 85.6461 0.984211779 98.42118

100 1 2 2 39.6 2.41 95.436 8.07 27.32 92.4885 0.969115694 96.91157

800 1 2 2 39.6 2.41 95.436 8.07 27.35 92.6918 0.971245228 97.12452

1500 1 2 2 39.7 2.42 96.074 8.07 27.44 93.3028 0.971155573 97.11556

4000 1 2 2 39.8 2.39 95.122 8.07 27.51 93.7794 0.98588594 98.58859

12000 1 2 2 39.8 2.12 84.376 8.07 25.61 81.2729 0.963222656 96.32227

20000 1 2 2 39.9 1.49 59.451 8.07 21.16 55.4827 0.933251352 93.32514 0.968298318

Page 140: DSP Modulated Class D Audio Amplifier

129

Input Freq.

[Hz]

Mod. Freq.

[MHz]

Mod.

Order

Input

Amplitude

Vin

[Volts]

Iin

[Amps]

Pin

[watts]

Rout

[Ohms]

Vout

(RMS)

Pout

[watts] Efficiency

Efficiency

[%] Average Efficiency

20 2 5 1.5 39.8 1.38 54.924 8.07 20.55 52.3299 0.952769748 95.27697

100 2 5 1.5 39.8 1.39 55.322 8.07 20.63 52.7382 0.953294416 95.32944

800 2 5 1.5 39.9 1.38 55.062 8.07 20.59 52.5338 0.954085238 95.40852

1500 2 5 1.5 39.8 1.38 54.924 8.07 20.53 52.2281 0.950916111 95.09161

4000 2 5 1.5 39.9 1.37 54.663 8.07 20.45 51.8219 0.948024644 94.80246

12000 2 5 1.5 39.9 1.24 49.476 8.07 19.45 46.8776 0.947482278 94.74823

20000 2 5 1.5 39.9 0.95 37.905 8.07 16.93 35.5173 0.937009255 93.70093 0.949083099

20 2 4 1.5 39.8 1.37 54.526 8.07 20.5 52.0756 0.955059762 95.50598

100 2 4 1.5 39.8 1.36 54.128 8.07 20.47 51.9233 0.95926847 95.92685

800 2 4 1.5 39.8 1.37 54.526 8.07 20.42 51.6699 0.947620182 94.76202

1500 2 4 1.5 39.8 1.37 54.526 8.07 20.49 52.0248 0.954128224 95.41282

4000 2 4 1.5 39.8 1.37 54.526 8.07 20.53 52.2281 0.957857105 95.78571

12000 2 4 1.5 39.8 1.24 49.352 8.07 19.13 45.3478 0.91886487 91.88649

20000 2 4 1.5 39.8 0.94 37.412 8.07 16.33 33.0445 0.88325867 88.32587 0.939436755

20 2 3 1.5 39.8 1.38 54.924 8.07 20.57 52.4318 0.95462519 95.46252

100 2 3 1.5 39.8 1.37 54.526 8.07 20.7 53.0967 0.973785979 97.3786

800 2 3 1.5 39.8 1.37 54.526 8.07 20.54 52.279 0.958790461 95.87905

1500 2 3 1.5 39.8 1.39 55.322 8.07 20.59 52.5338 0.949601269 94.96013

4000 2 3 1.5 39.8 1.39 55.322 8.07 20.53 52.2281 0.944074988 94.4075

12000 2 3 1.5 39.8 1.24 49.352 8.07 19.41 46.685 0.945960014 94.596

20000 2 3 1.5 39.8 0.91 36.218 8.07 16.73 34.6831 0.957621488 95.76215 0.95492277

20 2 2 1.5 39.8 1.37 54.526 8.07 20.56 52.3809 0.960658537 96.06585

100 2 2 1.5 39.8 1.38 54.924 8.07 20.54 52.279 0.951842704 95.18427

800 2 2 1.5 39.8 1.39 55.322 8.07 20.5 52.0756 0.941317895 94.13179

1500 2 2 1.5 39.8 1.37 54.526 8.07 20.52 52.1772 0.956924203 95.69242

4000 2 2 1.5 39.8 1.37 54.526 8.07 20.43 51.7206 0.948548539 94.85485

12000 2 2 1.5 39.8 1.2 47.76 8.07 19.15 45.4427 0.951480087 95.14801

20000 2 2 1.5 39.8 0.87 34.626 8.07 16.25 32.7215 0.944997961 94.4998 0.950824275

20 2 1 1.5 39.8 1.25 49.75 8.07 19.56 47.4094 0.952952121 95.29521

100 2 1 1.5 39.8 1.25 49.75 8.07 19.53 47.2641 0.950031197 95.00312

800 2 1 1.5 39.8 1.25 49.75 8.07 19.57 47.4579 0.953926759 95.39268

1500 2 1 1.5 39.8 1.27 50.546 8.07 19.54 47.3125 0.936027894 93.60279

4000 2 1 1.5 39.8 1.26 50.148 8.07 19.47 46.9741 0.936709125 93.67091

12000 2 1 1.5 39.8 1.13 44.974 8.07 18.39 41.9073 0.931812234 93.18122

20000 2 1 1.5 39.8 0.84 33.432 8.07 15.46 29.6173 0.885896705 88.58967 0.935336576


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